| 00:00.05 | F2Knight | caller --- calls docotor --- doctor transfers to ---- asterisk --- sets appointment --- specail app |
| 00:00.12 | mathi | WIMPy, you told earlier you were not sure |
| 00:00.44 | F2Knight | WIMPy, no. if I call you and you transfer my call to someone elase that someone else will not get both numbers. esp when its over pots |
| 00:00.50 | WIMPy | I am not sure if ITSPs will give you the information (and if they do if Asterisk understands it) but the PSTN definitely does. |
| 00:00.50 | ChannelZ | I thought the point was not to have to talk to the doctor ot make an appointment |
| 00:01.09 | ChannelZ | the doc having to transfer makes it more complicated |
| 00:01.12 | WIMPy | Sure you get both numbers. |
| 00:01.19 | F2Knight | docotor-transfer would be an ivr (hopefully) |
| 00:01.26 | mathi | ChannelZ, I think we are talking about automatic transfer (if i'm not wrong) |
| 00:01.28 | WIMPy | Plus the reason why the call was forwarded to you. |
| 00:02.02 | F2Knight | no WIMPy, you would only get the incoming callerID |
| 00:02.10 | mathi | WIMPy, but if there is an intermediate entity at the doctor's place, I might lose that information |
| 00:02.15 | WIMPy | mathi: You want the caller do directly reach your IVR without someone at the doctors office talking to them first? |
| 00:02.29 | ChannelZ | My understand of what you wanted was: caller calls doctors phone number but gets an IVR -- makes an appointment through your app, or presses 1 (whatever) to actually talk to the doctor |
| 00:02.35 | WIMPy | F2Knight: Have you used the PSTN in the last 20 Years? |
| 00:02.37 | mathi | WIMPy, yes, the whole point is to not disturb the doctor |
| 00:02.40 | F2Knight | mathi, exactly.. that is why you can provide a prompt to collect the callers phone number |
| 00:02.59 | WIMPy | mathi: Ok, so you want forwarding, not transfer. |
| 00:03.16 | mathi | F2Knight, yes, but if I install a server at each doctor's workplace, I don't have to ask that. now of course it's expensive for that extra feature |
| 00:03.28 | mathi | but it is annoying to type in the phone number |
| 00:03.32 | F2Knight | forwarding the call yes. and as such you would not get that callerid passed from the real caller |
| 00:04.02 | mathi | that's annoying |
| 00:04.08 | WIMPy | F2Knight: Sto that b***, please. The PSTN WILL do that! |
| 00:04.16 | F2Knight | mathi, if you install an server at each doctors work place you will have to install additional equipment at each doctors work place and maintain said equipment |
| 00:04.35 | mathi | F2Knight, I know |
| 00:05.04 | WIMPy | The PSTN supports a lot of caller IDs. |
| 00:05.10 | WIMPy | More than Asterisk. |
| 00:05.46 | mathi | WIMPy, I think F2Knight wanted to say that after the forward, the number of the patient will be lost (the pstdn info will be lost), instead the caller will be the doctor's number which will forward to the server |
| 00:05.53 | F2Knight | WIMPy, please prove me wrong on that. If I pick up my desk phone and call you, and have you send forward that call to my asterisk box I will not beable to tell the number from my deskphone, and your phone number that you forwarded the call to me from |
| 00:06.03 | WIMPy | mathi: Yes, but that's wrong. |
| 00:06.41 | F2Knight | mathi, considering you are not knowledgable about phones I do not think this is a very effective way fo ryou to be spending your resources. as it will incure a higher cost per installation. |
| 00:06.54 | F2Knight | ther eare ways to work around it that are much more cost effective |
| 00:06.54 | WIMPy | If you call someone who has set up forwarding to me and you call that other persons number, I get both your number and the forwarding persons number on my display. |
| 00:07.33 | WIMPy | And likewise the caller will get my number on his display. |
| 00:07.50 | F2Knight | only one number would show up on the display |
| 00:07.53 | F2Knight | not 2 |
| 00:08.21 | WIMPy | You get both numbers. |
| 00:08.32 | ChannelZ | has no idea what this conversation is even about now |
| 00:08.38 | WIMPy | At least in Europe. |
| 00:09.17 | F2Knight | Caller --> POTS <--> POTS <-- Doctor office ---> POTS < ---> DID <--- ASTERISK ${CALLERID(all)} = Doctors POTS |
| 00:09.19 | WIMPy | It's about the need for both the callers number and the forwarders number. And that's not an issue at all. |
| 00:09.48 | ChannelZ | doesn't know how the forwarding got involved |
| 00:09.55 | F2Knight | Its not a forward.. its a transfer. |
| 00:10.09 | mathi | F2Knight, is it possible to briefly conclude what I need in the case I centralize and host it, using ITSP? I need to rent several incoming channels to my asterisk server right? and link X channels to each doctor's number? |
| 00:10.26 | WIMPy | CALLERID(num) should give the patient and CALLERID(RDNIS) the doctor. |
| 00:10.28 | F2Knight | it must be a transfer because the caller might have wanted to call the doctor for something else. |
| 00:11.17 | F2Knight | mathi, Channels are NOT phone numbers.. Channels are the number of calls you can accept over a single 'number' a number is called a DID |
| 00:11.27 | mathi | F2Knight, I think I need to forward the call to my server, and if he choose "urgent demand", he will be transferred to doctor phone |
| 00:11.31 | ChannelZ | Caller -> ITSP -> Master Server -> (caller makes appointment, or wants to talk to doctor) -> Master Server calls doctor |
| 00:11.46 | WIMPy | F2Knight: No it's the number of _active_ call you can have. |
| 00:11.59 | F2Knight | mathi, that is the wrong way to handle it becuase now you have to handle all the doctors calls for them. |
| 00:12.21 | mathi | F2Knight, but that's what I suppose the doctor wants |
| 00:12.32 | mathi | their public number is the number to take appointment |
| 00:12.42 | mathi | and that's mainly it |
| 00:12.47 | F2Knight | what else is that number for? |
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| 00:12.56 | mathi | nothing |
| 00:13.00 | F2Knight | is it at all ever used to contact the doctor? |
| 00:13.09 | F2Knight | billing? |
| 00:13.15 | F2Knight | general questions? |
| 00:13.16 | mathi | only in urgent demand |
| 00:13.17 | mathi | no |
| 00:13.33 | F2Knight | how do you get ahold of the doctor? |
| 00:13.37 | F2Knight | a different number? |
| 00:13.58 | mathi | I can still add a menu in IVR |
| 00:14.12 | ChannelZ | they should give out the 'real' number if they want |
| 00:14.22 | mathi | all calls to that number will automatically go to IVR, without intermediate receptionnist |
| 00:14.40 | mathi | it's a forward |
| 00:15.01 | F2Knight | if this 'appointment' number is only for apointments. then how is this any different then my original statement of providing a number for each doctor just for appointments |
| 00:15.07 | ChannelZ | "Our main number is 555-111-2222 to make appointments. For urgent matters, press 1 now or call 555-333-4444" |
| 00:15.42 | F2Knight | that only works if every doctors office has a IVR that can be programed. |
| 00:15.49 | ChannelZ | we're turning a simple doctors appointment into neurosurgery |
| 00:15.57 | mathi | F2Knight, i thought you meant one number for all doctors sorry. so how I implmeent this, |
| 00:15.58 | F2Knight | which might not be the case at all , as mathi said not everyone has a phone sytem |
| 00:16.15 | ChannelZ | They don't need a phone system. |
| 00:16.26 | WIMPy | That's what I already said. |
| 00:16.38 | F2Knight | ChannelZ, how do you propse to play an IVR for the caller? |
| 00:16.46 | ChannelZ | Asterisk does it |
| 00:17.00 | F2Knight | Then mathi has to be in the middle of all the calls again. |
| 00:17.05 | WIMPy | You forward that number to mathi. |
| 00:17.09 | ChannelZ | The doc has a phone number already presumably. That gets ported to mathi's system for taking automated appointments. |
| 00:17.35 | F2Knight | sure. so mathi has a did to take the calls on he now has to host IVR's for each doctor. |
| 00:17.59 | ChannelZ | If the person wants to talk to the office directly, he can either be middle man and place the outgoing call to them as a convenience, or optionally the doc gives out his "real" number (or the IVR tells them what it is that they can call themselves) |
| 00:18.06 | WIMPy | You need to check local regulations. No idea if NP would be an option. |
| 00:18.31 | F2Knight | and bridge calls back out to the doctor. if his VPS server is over loaded , is off line or has some other issue, then the URGENT care option never gets reached. |
| 00:18.57 | mathi | WIMPy, NP? |
| 00:18.57 | ChannelZ | Or if a drunk plows into the phone poll, the doc has no phone service for a week. |
| 00:19.03 | ChannelZ | Number Porting |
| 00:19.22 | WIMPy | ^ |
| 00:19.47 | ChannelZ | I mean there's 10 different ways to do this depending on how transparent you want it to be. Or how little the doctor actually does or doesn't want to talk with people. |
| 00:20.27 | F2Knight | ChannelZ, that is true.. but I think there are good odds his server will have issues or bandwidth or config problems more often then the POTS going down.. esp considering that POTS in most places in the US are not on 'telephone' polls and are run underground |
| 00:20.29 | mathi | F2Knight, is there any solution in the case the VPS server is down temporarily ? |
| 00:20.33 | ChannelZ | It can be as simple as an alternate phone number for appointments only that the office tells their clients to call in the future. They still have the 'regular' number if they know they want to talk to a human. |
| 00:20.45 | ChannelZ | Forget forwarding or transferring or any bullshit at all |
| 00:21.01 | F2Knight | mathi, a backup server and High Availablity setup in a small cluster |
| 00:21.17 | F2Knight | if one VPS goes down it could reroute to another |
| 00:21.42 | F2Knight | ChannelZ, that is what I suggested about an hour ago :) |
| 00:22.13 | mathi | the primary number is the number to make appointments |
| 00:22.18 | ChannelZ | fine |
| 00:22.19 | WIMPy | Or just forward the call back in case of failure. |
| 00:22.21 | F2Knight | and if they have an IVR feature they can have it automaticly just dial in. and if its receptionist based she can transfer |
| 00:22.25 | ChannelZ | That one terminates at your server. |
| 00:22.30 | F2Knight | that way it works with everyone. |
| 00:22.48 | F2Knight | and he has only to worry about his core business, the app, and not about phone services |
| 00:25.09 | ChannelZ | It's still the 'transferring' part that adds complexity back in the case of these doctors offices using POTS |
| 00:27.38 | ChannelZ | If it were me, from what I can tell the needs/wants of the doctors are, I'd have their main number (or whatever) go to my system running somewhere, do the IVR<->calendar integration as necessary, and either direct people to call a DIFFERENT number to talk to the office directly or burn a channel and connect them myself |
| 00:27.40 | WIMPy | Why do you think they could have POTS? |
| 00:28.07 | ChannelZ | The impression I got from way earlier was these offices were ghetto and didn't already have a PBX of any sort or even multiple phone lines |
| 00:28.18 | WIMPy | agrees |
| 00:28.45 | WIMPy | Maybe not, but they will most probably have BRIs. |
| 00:29.14 | WIMPy | Doctors tend to uses faxes. |
| 00:30.03 | ChannelZ | Without more concrete info I'm done guessing and doing What Ifs and making suggestions |
| 00:31.50 | mathi | ChannelZ, yes sure, how do you get their main number go to the remote system |
| 00:32.29 | WIMPy | Anyway. They will surely want internal transfers, so they will have PBXes as well. |
| 00:32.47 | ChannelZ | here in the US you port the number to your ITSP if you're doing it VoIP or to your telco if you're going to physically connect yourself. |
| 00:33.19 | WIMPy | PBXes can make the thing a little more difficult however. |
| 00:34.02 | WIMPy | They usually intercept transfer reequests to do them themselves instead of just forwarding the request to the switch. |
| 00:34.12 | WIMPy | But that's just a matter of finding the right code. |
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| 01:51.30 | SeRi | ? |
| 02:10.11 | beccara | is anyone aware of a reason why when a call is terminated by the initating party under 1.8.7 under a macro the h context is not hit? |
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| 04:03.59 | bluregard | is there a way to have asterisk send channel details such as number dialed, call duration etc via AMI when using originate? |
| 04:07.06 | ChannelZ | send where? |
| 04:08.34 | bluregard | back to the manager session |
| 04:08.39 | Kobaz | bluregard: you would have to write that yourself |
| 04:08.50 | Kobaz | the ami will send you a newchannel event, and a hangup |
| 04:09.01 | Kobaz | and you can calculate the difference in time to get the call duration |
| 04:09.06 | Kobaz | or hangup - pickup time to get talk time |
| 04:09.15 | Kobaz | you could also look at the cdr log |
| 04:10.11 | bluregard | so what's the point of the actionID? I figured that was a way to keep track of actions that were handed to * in order to check on their status. |
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| 04:30.07 | bluregard | according to http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+Originate I should be getting a lot more in response to the originate than I am. |
| 04:30.47 | bluregard | all I get is the "Response:", none of the "Event:"s |
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| 04:38.39 | SeRi | waz up guys |
| 04:39.25 | bluregard | o/ |
| 04:40.20 | SeRi | \o |
| 04:40.51 | SeRi | time for some caffeine.... :) |
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| 05:12.13 | ChannelZ | bluregard: what auth does your manager user have? |
| 05:17.00 | bluregard | I'm using read/write all |
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| 05:26.55 | bluregard | shouldn't a sip user connecting/disconnecting or issueing a core reload generate an AMI event? |
| 05:30.09 | carrar | moo |
| 05:35.09 | ChannelZ | hmm if a software raid1 in linux is resyncing, will rebooting make it start from 0% again? |
| 05:36.36 | carrar | and take the chance that you only drive thats usable could go tits up also? |
| 05:36.44 | carrar | I'd let it sync |
| 05:37.15 | ChannelZ | well something is wrong, if I pause the resync and do some hdparm -t tests, one of the disk returns wildly varying numbers |
| 05:38.04 | ChannelZ | as low as 300k/sec up to 38MB/sec, which would explain why the resync speed is going up and down and even at low limits is bogging the whole machine down |
| 05:38.05 | carrar | might be a good time to make a backup :) |
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| 06:00.04 | gajini | Hi, Could you tell me, how to install openr2 with asterisk 1.4 and Asterisk 1.2? |
| 06:03.37 | irroot | gajini there patches for it but you will be better off using 1.8 |
| 06:04.22 | irroot | carrar mondays are always good for backup excellent way of dodging meetings while been responsible .... |
| 06:04.54 | irroot | ChannelZ lo there hope it comes right |
| 06:05.40 | gajini | ok, i am using PRI card, which is using zaptel driver, So i have to use asterisk 1.4 or asterisk 1.2, |
| 06:06.18 | irroot | gajini well you could update to dahdi ? |
| 06:07.39 | gajini | irroot, thanks . This card is using tor3e driver with zaptel, which is supporting only asterisk 1.4. versions , Any other way to do |
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| 06:09.04 | gajini | irroot , i dont have dahdi driver for my card, so i am using this older zaptel driver only |
| 06:09.10 | irroot | gajini you got the asterisk source code ?? and all the build tools ?? build asterisk to make sure it builts |
| 06:09.17 | irroot | builds |
| 06:09.50 | irroot | then install openr2 and get the patch on the openr2 site |
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| 06:12.19 | gajini | irroot, i tried openr2-1.3.2.tar.gz with asterisk-1.2.31.1 , its compiled and installed. But i didnt get mfcr2 application in asterisk console. Could u help me please |
| 06:12.56 | irroot | gajini did you patch asterisk with the right patch ? |
| 06:13.47 | irroot | and run ./bootstrap.sh in the root of asterisk source |
| 06:14.03 | bluregard | is there any way to relate the UniqueID: in AMI events with the Action: that initiated said event? |
| 06:14.08 | gajini | irroot, i installed this patch - asterisk-1.2.31.1-patch |
| 06:16.33 | gajini | irroot, it seems this asterisk version not detected openr2 |
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| 06:17.40 | irroot | gajini you need openr2 installed first then patch asterisk |
| 06:17.59 | irroot | once this is done with no errors run ./bootstrap.sh then configure |
| 06:19.03 | gajini | irroot, i installed openr2 first, and i apply patch for asterisk, then i installed asteisk |
| 06:19.43 | gajini | irroot, i couldnt see that ./bootstrap.sh file in my source |
| 06:20.56 | irroot | gajini you need to update the configure script to build it on 1.4+ its bootstrap.sh maybe autoreconf will work on 1.2 |
| 06:21.39 | gajini | irroot, thanks. let me try this |
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| 07:31.56 | schmidts | good morning |
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| 08:07.13 | beccara | does anyone know of a way to output dumpchan into a odbc command for storing debugging? |
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| 08:17.10 | jacc0 | morning |
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| 08:26.21 | dom| | why does an incoming call to "agent-11" oder "agent11" not match on an extension named "_agent."? |
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| 09:31.23 | mandla | http://pastebin.com/YrJ6WhYt |
| 09:31.48 | mandla | Hello guys, im having a problem with call transfering. |
| 09:32.39 | mandla | Can you please help, the url is for a code snippet for call transfer in features.conf. |
| 09:33.28 | irroot | mandla hi there been bit hectic this morning need to see the extensions.conf and call trace "core set verbose 3" |
| 09:33.50 | irroot | when you make the call and attempt to press # |
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| 09:34.37 | ilj | Hi! What does forward slash mean in Background application, like in this: Background(hellouser/day) ? |
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| 09:42.14 | mandla | irroot, http://pastebin.com/xrvgzeuY |
| 09:42.25 | mandla | irroot, extension.conf |
| 09:45.25 | mandla | irroot, http://pastebin.com/0Y9YiUqL |
| 09:45.48 | *** join/#asterisk Rico29 (~rico@oceanet-telecom-fttb-129-2.olm.fr) |
| 09:46.40 | irroot | mandla you must use the t/T options to enable the codes in the dial application |
| 09:47.08 | mandla | irroot, tzafrir , where exactly?? |
| 09:47.48 | mandla | irroot, tzafrir, was thinking im already a master of Asterisk, lol, i was wrong. |
| 09:47.49 | irroot | Executing [917@default:1] Dial("DAHDI/1-1", "DAHDI <- in the [default] section in extensions.conf |
| 09:47.55 | Rico29 | I've got a little problem with MoH and realtime... |
| 09:48.00 | Rico29 | asterisk -r |
| 09:48.02 | Rico29 | oops |
| 09:48.21 | tzafrir | mandla, you seem to have there native bridging. The call does not pass through Asterisk |
| 09:48.28 | Rico29 | when I do a "moh show classes", I don't see my realtime classes |
| 09:48.38 | tzafrir | How would asterisk be able to detect digits for transfer? |
| 09:48.41 | Rico29 | just the default class configured in musiconhold.conf |
| 09:49.43 | tzafrir | You didn't even ask Asterisk to look for it. You have: Dial(DAHDI/30). No options (the third field), Specifically: no 't' |
| 09:49.49 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 09:50.06 | mandla | tzafrir, what do you mean it doesnt pass through asterisk?? I thought im using asterisk. |
| 09:50.54 | tzafrir | You didn't ask Asterisk to be able to use transfer. And there was no other good reason. |
| 09:51.06 | irroot | mandla native bridging is when the call goes through the hardware directly and does not go via asterisk or atleast the audio does not |
| 09:51.08 | *** join/#asterisk Defraz (~T0tal@70.36.76.167) |
| 09:51.13 | tzafrir | Thus Asterisk optimized the call to go directly between the two DAHDI channels |
| 09:51.40 | Rico29 | can anyone take a look ? http://pastebin.com/eK6Y6ist |
| 09:51.41 | Rico29 | thanks |
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| 09:53.13 | tzafrir | Rico29, one minor thing: you demonstrated that the root mysql user can read that config. |
| 09:53.34 | mandla | tzafrir, irroot, that should be specified in features.conf not in extensions.conf |
| 09:53.35 | Rico29 | yes, and ? |
| 09:53.36 | tzafrir | But what about the mysql user Asterisk uses |
| 09:53.55 | tzafrir | mandla, Dial(DAHDI/30,,t) |
| 09:53.56 | Rico29 | tzafrir, > asterisk can read it too |
| 09:54.22 | Rico29 | works for realtime sip peers, realtime sip queues, ... |
| 09:54.39 | tzafrir | I'm not really sure 'moh show files' shows realtime. But I'm not really familiar with that |
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| 10:01.21 | mandla | irroot, how do i make it go via asterisk, is this not what i have been doing all along?? |
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| 10:06.10 | kaldemar | mandla: tT options in Dial or directmedia=no in sip.conf will make it go through asterisk. just use the Dial options. |
| 10:06.23 | Rico29 | tzafrir, even when I make a call with MoH to a realtime peer with 'cmapub' moh class, it plays 'default' class |
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| 10:35.50 | ilj | I'm trying to conjure up a simple auto attendant menu, and I'm thinking about using WaitExten() followed by GotoIf that checks $EXTEN variable (assuming it gets changed after WaitExten() is through) to send calls to different queues. Does this sound ok to you guys? I'm a complete newbie so bear ... |
| 10:35.55 | ilj | ... with me xD |
| 10:44.15 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 10:44.46 | kaldemar | if EXTEN changes, so will the extension. don't use it to store anything. |
| 10:44.49 | Rico29 | tzafrir, > fixed |
| 10:45.01 | puzzled | morning |
| 10:45.16 | Rico29 | it came from my database, moh field name has changed, moved from 'musiconhold' to 'mohsuggest' |
| 10:45.37 | kaldemar | ilj: use some other temporary variable name of your choice to hold the user-entered value. |
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| 10:49.59 | kaii | can asterisk pass a SIP notify originating from one peer to another peer? (in that case, act as a "proxy"?) |
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| 10:57.25 | ilj | kaldemar, could you please show how I could store the user entered exten? |
| 10:58.44 | kaldemar | ilj: with app Read |
| 11:00.02 | kaldemar | with WaitExten the user would immediately go to the new extension. if you want to do some kind of checks for the value, first read it to a variable with Read and then goto if it satisfies your requirements. |
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| 11:02.13 | ilj | hmm ok, thanks! |
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| 11:58.21 | mandla | irroot, |
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| 12:15.00 | dom| | why does an incoming call to "agent-11" or "agent11" not match on an extension named "_agent."? |
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| 12:25.07 | wdoekes2 | dom|: because n means [0-9] |
| 12:25.26 | wdoekes2 | dom|: _age[n]t. would work |
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| 12:26.17 | dom| | ah ok |
| 12:26.43 | wdoekes2 | *means 2-9 actually |
| 12:26.51 | wdoekes2 | or 1-9 |
| 12:27.17 | wdoekes2 | there's X=0-9, Z=1-9, N=2-9, iirc |
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| 12:56.07 | FlashDeluxe | hi! does anybody know a good tutorial for faxing over ISDN? I got a 4 bri card from junghans and asterisk 1.8 with dahdi installed |
| 12:58.38 | cVsup | somebody can say if Shaun works at digium? |
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| 13:05.27 | WIMPy | FlashDeluxe: The only difference to POTS is that you need to set the BC ("transfercapability"). |
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| 13:14.51 | jacc0 | is there a way to convert decimal to hex in dialplan? |
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| 13:18.17 | irroot | jacc0 with out using System :P |
| 13:18.43 | irroot | you could write a macro to process the digits one by one with math |
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| 13:19.25 | jacc0 | hmm, |
| 13:19.26 | irroot | or hopefully something like 0xFF is acceptable to the math funcs |
| 13:20.01 | jacc0 | okay, I made something like that in first grade ; I think I can do it again :P |
| 13:20.57 | jacc0 | Hmm, my second though -> maybe I coud use sql to convert |
| 13:21.01 | jacc0 | :P |
| 13:21.47 | jacc0 | hehehe, select HEX(123) , that will do :P |
| 13:22.04 | kaldemar | or just read the MATH documentation first. |
| 13:22.16 | wdoekes2 | jacc0: SPRINTF |
| 13:23.07 | jacc0 | @kaldemar: good idea |
| 13:24.27 | FlashDeluxe | WIMPy: that was not my question ;-) I`d like to install the fax-server regarding to a good tutorial and i asked if anybody does know one ;-) |
| 13:25.08 | jacc0 | @wdoekes2: I didn't know it was in asterisk, I'll look into that also |
| 13:25.25 | wdoekes2 | didn't either, but a core show functions revealed it to me |
| 13:26.11 | WIMPy | FlashDeluxe: I just said that you don't need an ISDN specific tutorial, except for an extra line in your dialplan. |
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| 13:27.37 | wdoekes2 | :n |
| 13:31.19 | fireman_biff | I have a Grandstream GXP2000 that I've tested on 2 PBXs. On one PBX the provisioning works fine but on the other it doesn't, although both PBXs seem to be setup similarly. On the one that doesn't work I can see that dhcpd is telling the phone the correct tftp-server-name, but then no requests come in for config files. Any advice on how I should troubleshoot? (I'm using elastix and the config files were automatically generated, but this seems to be a pro |
| 13:32.43 | WIMPy | How could it request configs if you don't tell it fromwhere to request them? |
| 13:34.37 | fireman_biff | WIMPy: dhcpd is telling the phone the correct tftp-server-name |
| 13:35.13 | fireman_biff | its set to tftp://<ipaddress> |
| 13:35.22 | fireman_biff | just like on the PBX that works |
| 13:35.24 | WIMPy | Oops. Sorry.Misread that. Does the name resolve? Is the domain correct? |
| 13:35.42 | fireman_biff | its just an IP, nothing to resolve |
| 13:36.12 | WIMPy | In the same subnet? Ist the netmask correct? |
| 13:36.46 | fireman_biff | domain is asterisk.local on both boxes, everything IP related seems fine, the phone gets on the network and can be pinged from the PBX |
| 13:36.54 | fireman_biff | if I set the account info manually it logs in |
| 13:37.50 | fireman_biff | I assume it doesn't make a difference to tftp, but the phones are on a separate VLAN from the data |
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| 13:38.06 | fireman_biff | but like i said, the phone and PBX see each other, and manual config works |
| 13:39.38 | WIMPy | And the tftp server? |
| 13:41.08 | fireman_biff | seems fine, I tried to download a file with microsofts tftp client and although it timed out, i at least saw the request in the PBX's /var/log/messages, which is the same thing that happened when I tried it against the PBX that can do provisioning |
| 13:41.52 | fireman_biff | its definitely listening on the default port 69 on all IPs |
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| 13:50.13 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 16:34.34 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 16:34.34 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta2 (2011/09/27), 1.8.7.1 (2011/10/17), dahdi-linux 2.5.0.2 (2011/10/21), dahdi-tools 2.5.0.2 (2011/10/21), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 16:34.36 | ChannelZ | missed the link before but no matter, scampers off to work... |
| 16:34.53 | wcselby | mathi read the description of what's happening in the verbose statements |
| 16:34.59 | wcselby | we're checking the hotdesk status |
| 16:35.42 | *** join/#asterisk cVsup (~cVsup@201.78.47.2) |
| 16:35.55 | mathi | wcselby, but there is a SET which is done for each execution of this extension. If the variable would be just EXT_STATUS for example, I'm not sure there would be any problem |
| 16:35.57 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 16:36.00 | cVsup | how can connect gsm interface to fxo port? |
| 16:36.16 | mathi | (referring to same => n,Set(${E}_STATUS=${HOTDESK_INFO(status,${E})})) |
| 16:36.20 | wcselby | mathi but since you're pattern matching, you need to be able to specifiy which extension you're looking up the status for |
| 16:36.29 | WIMPy | cVsup: Use a gateway |
| 16:36.35 | wcselby | since you could be dialing 1101, or 1105, or whatever |
| 16:36.36 | r0m|u | cVsup, what type of gsm interface? |
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| 16:36.48 | [TK]D-Fender | mathi, Yes it looks pointless for those variable names to be variable. |
| 16:36.55 | mathi | wcselby, yes, you can have this information with ${E} |
| 16:36.59 | r0m|u | cVsup, I do what WIMPy recommended |
| 16:37.14 | [TK]D-Fender | mathi, Now find something new to neurose about :p |
| 16:37.15 | mathi | wcselby, I'm talking about naming variable ${E}_STATUS, which seems to be useless here |
| 16:37.19 | cVsup | I need connect gsm interface to fxo port. Incoming calls work but outgoing nothing, somebody can help? |
| 16:37.22 | Qwell | It's a book. A book that explains how things work. Arguing over syntax (when it clearly explains the syntax on the next paragraph) is silly. |
| 16:37.23 | WIMPy | That's the only way if it is to be connected to an FXO. |
| 16:37.37 | [TK]D-Fender | mathi, It is useless. Time to move on now :) |
| 16:37.39 | wcselby | mathi well, that's the thing about asterisk, you can do the same thing 50 different ways. do it the way you like. :) |
| 16:37.58 | r0m|u | cVsup, you neetd a gsm gateway |
| 16:39.16 | mathi | ChannelZ, so would you install a server at each workplace or use a centralized server? |
| 16:39.23 | mathi | ChannelZ, ok i'm joking ))))) |
| 16:39.27 | r0m|u | cVsup, and like WIMPy said I dont think is possible to route outbound calls to the gsm threw FXO. a gateway will act as a sip device and allow incoming and outgoing calls |
| 16:39.35 | *** part/#asterisk _N1x (~n1x@95.104.13.145) |
| 16:40.15 | WIMPy | There are GSM<>SIP, GSM<>POTS, GSM<>BRi and even GSM<>PRI gateways available. |
| 16:40.36 | WIMPy | You just need to pick the right one. |
| 16:40.54 | mathi | ChannelZ, but seriously, I was looking for ISTP who provides a good amount of incoming channels (for simultaneous incoming calls), and it's very ahrd to find one for a good price |
| 16:41.00 | mathi | *ITSP |
| 16:41.27 | WIMPy | mathi: How many channels do you want? |
| 16:41.39 | [TK]D-Fender | WIMPy, BELGIUM |
| 16:41.48 | WIMPy | I know |
| 16:42.13 | WIMPy | We had the issue of american advice tonight. |
| 16:42.16 | [TK]D-Fender | WIMPy, Just making sure, because very little critical info tend to be flowing here lately :) |
| 16:43.30 | mathi | WIMPy, well I was looking for a flexible provider who allows me to pay extra channels for a decent price, because I wish to start small but the system may grow. I'm talking about hundreds of channels |
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| 16:44.08 | WIMPy | Hundreds of simultaneous calls??? Are you going to offer that service worldwide? |
| 16:44.38 | mathi | no, ok probably I need less :-) |
| 16:45.17 | [TK]D-Fender | mathi, Weren't we starting with having only one single person who wanted to work with you? |
| 16:45.37 | mathi | [TK]D-Fender, yes, but then I need to foresee more customers |
| 16:45.43 | mathi | 1, 10, 100, ... |
| 16:45.48 | [TK]D-Fender | mathi, And you were about to get a single analog line and didn't want to shell out for card for any devices to plug it? Or buy SIP phones for use on desks? |
| 16:46.36 | [TK]D-Fender | mathi, 1 -> 100. Your margin of error is astounding and makes giving you advice dubious. Your answers and needs change too much too fast. |
| 16:46.54 | WIMPy | Get a quad BRI card, connect one BRI and up to three more if you need them. |
| 16:47.17 | [TK]D-Fender | mathi, Yuo need to be be clearer and more realistic over the terms of your goals |
| 16:47.41 | WIMPy | Yes, we ar not that good about dreams :-) |
| 16:48.06 | mathi | [TK]D-Fender, it's simple, I have one customer now, using another soft, and he wants to try that IVR, the goal is to implement it for 100 doctors, what is unrealistic ? |
| 16:48.07 | [TK]D-Fender | mathi, Otherwise you will get potentiall costly advice and not scal to where you need it when you need it and make correcting for the improper planning very costly and painful |
| 16:48.25 | [TK]D-Fender | mathi, "when" is this goal? |
| 16:48.35 | mathi | in a year |
| 16:48.41 | WIMPy | It's like out local heroes. The all start with ordering a PRI, even if they wouldn't ever had a beed for mot than 3 BRIs. |
| 16:48.50 | [TK]D-Fender | then start with a provider that can already scal to where you need |
| 16:49.07 | *** join/#asterisk hacim (~micah@debian/developer/micah) |
| 16:49.15 | WIMPy | mathi: But 100 doctors is not the same as 100 simultaneous calls! |
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| 16:49.32 | mathi | WIMPy, I know :) |
| 16:49.48 | mathi | WIMPy, but 100 doctors have 1000+ patients |
| 16:49.59 | hacim | can anyone recommend a DiD provider that i can use for one measly 30second outgoing call a month? I dont want to pay high monthly obviously for this purpose |
| 16:50.03 | WIMPy | Then sto telling us about your dreams where you have hundreds of simultaneous calls. |
| 16:50.05 | mathi | 10000+ sorry |
| 16:50.16 | [TK]D-Fender | mathi, And you are throwing "doctors" and "patients" around like they are telecom units of measure. |
| 16:50.22 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 16:50.30 | [TK]D-Fender | mathi, Trying to spec this out is becoming needlessly painful |
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| 16:50.41 | [TK]D-Fender | 10000 now? |
| 16:50.49 | [TK]D-Fender | This story gets better every time I hear it... |
| 16:50.59 | [TK]D-Fender | moves on to more productive matters |
| 16:51.03 | WIMPy | Millions of calls per second. |
| 16:51.28 | elliot98 | gives a friendly wave to all |
| 16:52.10 | WIMPy | Well, for that amounts I'd watch out for an used EWSD on ebay. |
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| 16:52.47 | mathi | [TK]D-Fender, I was telling WIMPy that each doctor have 1000+ patients, so for my goal of 100 doctor, YES, I may have a good amount of calls! Don't you agree? |
| 16:53.08 | hudony | Hi there : when defining my outbound trunk in extensions.conf, I am entering my username:password@host. Proble mis my username contains @ which invalidate the rest of the string |
| 16:53.14 | [TK]D-Fender | mathi, We went from 1..... to 100, to *10000*. |
| 16:53.17 | WIMPy | mathi: no |
| 16:53.24 | hudony | Does anybody have an idea how evade this problem? |
| 16:53.35 | [TK]D-Fender | mathi, Tracking what really matters in't proving to be worth the trouble |
| 16:53.41 | WIMPy | How often is each patient going to call? An average once per month perhaps? |
| 16:53.53 | WIMPy | I'd guess even less. |
| 16:53.57 | mathi | [TK]D-Fender, I was only talking about the number of patients that MAY call, I am conscious that I won't have more than 100 simultaneous calls... I just would like to foresee it |
| 16:54.12 | [TK]D-Fender | hudony, Make a proper peer and stop putting auth info into extensions.conf |
| 16:54.22 | libryder | lol |
| 16:54.40 | mathi | WIMPy, not much, listen let's forget about the 100 calls simultaneously :) |
| 16:54.58 | WIMPy | That sounds like a plan! |
| 16:55.15 | [TK]D-Fender | mathi, That is aout the only useful piece of information we could possibly use and you tell him to disregard it... |
| 16:55.36 | hudony | ok! |
| 16:55.38 | hudony | Thanks |
| 16:55.53 | mathi | [TK]D-Fender, because you were criticizing me so much about it |
| 16:56.10 | WIMPy | What do you think the average time they spend in the IVR would be like? |
| 16:56.46 | [TK]D-Fender | mathi, You throw out the one good piece of info. I'm irked by the 5 other numbers and superfluous "facts" you left in its place. |
| 16:56.47 | mathi | WIMPy, not more than a minute |
| 16:57.04 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 16:57.24 | WIMPy | Ok, with 100000 patients calling for one minute every month, that makes an average of 2.3 simultaneous calls. |
| 16:58.01 | WIMPy | Off course there won;t be many calling at night, so you'd need to be prepared for at least 4 simultaneous calls. |
| 16:58.15 | WIMPy | </realitycheck> |
| 16:58.31 | [TK]D-Fender | WIMPy, wannt play a game of Fizbin? |
| 16:58.46 | [TK]D-Fender | WIMPy, You're reality check ... just bounced. |
| 16:58.46 | WIMPy | Whazzat? |
| 16:58.53 | [TK]D-Fender | WIMPy, JFGI :) |
| 16:59.31 | libryder | we have peaks of 8 simultaneous calls/day and we definitely don't have 100,000 customers |
| 16:59.43 | mathi | ^ |
| 16:59.47 | timeshell | Anyone know if current versions of asterisk will compile on cygwin? |
| 16:59.49 | WIMPy | libryder: What average call time? |
| 16:59.56 | libryder | ~8 minutes |
| 17:00.10 | WIMPy | We're talking about an IVR with <1 minute. |
| 17:00.13 | [TK]D-Fender | WIMPy, http://www.youtube.com/watch?v=k0SsR2y6Tgo |
| 17:00.54 | mathi | WIMPy, so I could find a provider who gives me 4-5 channels, do you know any reliable provider in belgium? |
| 17:01.13 | WIMPy | I'd go for a real phone line. |
| 17:01.35 | WIMPy | That way you can be sure you get both caller and redirecting ID. |
| 17:01.51 | *** join/#asterisk mpe (~mpe@31.25.23.177) |
| 17:02.10 | mathi | WIMPy, that implies to put a server at the client's premises, (or I could host it at my own premises but that's too much of a hassle) |
| 17:02.18 | libryder | WIMPy: no matter how you do the math, 100 simultaneous calls is pretty out there for just about anything short of a ... i can't think of anything |
| 17:03.03 | WIMPy | mathi: Not at the client. Either in some place you have or in a data center. |
| 17:03.06 | libryder | i wonder how many calls a verizon call center would average |
| 17:03.42 | Qwell | libryder: 6 |
| 17:03.52 | mathi | WIMPy, you suggest me real phone line because SIP is not 100% reliable? |
| 17:04.29 | libryder | Qwell: is that a real number? |
| 17:04.34 | WIMPy | No, because I don;t think they will send you the information who forwarded the call to you. |
| 17:04.49 | WIMPy | And off course it is less reliable. |
| 17:05.31 | mathi | [TK]D-Fender, I realize 100 simultaneous calls was a stupid thing to say now that I estimate myself, I should have think twice. Don't be mad at me:p |
| 17:05.35 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 17:05.36 | libryder | mathi: this is who we use -- http://www.level3.com/ |
| 17:05.47 | libryder | (for sip) |
| 17:06.32 | hacim | does anyone have any good references for comparing DiD providers? |
| 17:06.53 | WIMPy | mathi: that's really the kind of think you should find out yourself before trying to figure possibilities to achieve it. |
| 17:11.11 | mathi | WIMPy, do you know what kind of server I would need ? |
| 17:12.44 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 17:13.10 | WIMPy | Next to nothing. |
| 17:14.29 | *** join/#asterisk ketas-ts (~ketas@82.131.22.194.cable.starman.ee) |
| 17:15.30 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 17:17.02 | r0m|u | mathi, does are questions that only you can answer. or hire a professional to do it for you. I am on the middle of hiring somebody to deploy a call center. I know where to draw the line. |
| 17:18.35 | r0m|u | I can play around all day with my setup at home but is not the same in real life production, I would not know where to begin :P |
| 17:18.54 | mathi | r0m|u, sure but now I have a more clear idea :) |
| 17:19.28 | r0m|u | Having a clear idea of how it works and asking what do you need are two different things. |
| 17:19.55 | libryder | elastix on an aws instance with a level3 trunk |
| 17:20.15 | *** join/#asterisk ekgermann (~ekgermann@204.87.224.41) |
| 17:20.22 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 17:20.23 | libryder | although i keep hearing running asterisk in the cloud is a bad idea |
| 17:20.46 | r0m|u | libryder, how is that working out for you? I tried elastix and it was way bloated... |
| 17:21.09 | r0m|u | I use to play with aws as a devel env for asterisk |
| 17:21.37 | libryder | r0m|u: we use it for out office phones so the it guys can manage the phones... it works pretty well for what we use it for |
| 17:21.49 | r0m|u | nice. |
| 17:22.54 | libryder | r0m|u: did you ever have any latency issues? |
| 17:23.24 | r0m|u | I moved everything in house due to issues in latency, etc... |
| 17:23.29 | r0m|u | libryder, yes |
| 17:23.45 | r0m|u | Thats why I moved everything in to a small devel box I built at home |
| 17:23.54 | *** part/#asterisk fireman_biff (~biff@65.48.133.103) |
| 17:24.40 | *** join/#asterisk Micc (~Micc@c-24-17-253-27.hsd1.wa.comcast.net) |
| 17:24.57 | r0m|u | I also had routing issue but I think that was comcast |
| 17:25.28 | Micc | I've got a channel thats been up for 94 hours and channel request hangup won't work, I've tried it on both legs of the call. Is there any other way to force the channel to die? |
| 17:26.57 | elliot98 | ok, so here's my question, I'm using several "register =>" commands in sip.conf to register multiple accounts with a specific server. When I issue a Dial command, how does asterisk know to associate the account with the particular registration? |
| 17:29.26 | *** join/#asterisk umay (~chris@67-6-158-37.hlrn.qwest.net) |
| 17:29.54 | cusco | hi |
| 17:29.55 | cusco | http://forums.digium.com/viewtopic.php?t=66833 |
| 17:29.58 | WIMPy | elliot98: not |
| 17:29.59 | cusco | have you noticed? |
| 17:30.00 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 17:30.04 | cusco | mysqli - too many connections |
| 17:30.05 | cusco | ! |
| 17:30.08 | WIMPy | elliot98: You dial peers. |
| 17:31.26 | elliot98 | WIMPy: how do I configure the peers so they send out from the port that it was registered with |
| 17:31.28 | elliot98 | ? |
| 17:32.04 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 17:32.04 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 17:32.38 | mathi | WIMPy, if I put my server in a datacenter and using real phone lines I have to 1. find a data center in belgium (they are rare and expensive), 2. find a data center that will want to put phone lines?? In the case I have a server at my own preùmises, in terms of availability 24/7, maintenance, it might get quite expensive I think |
| 17:33.03 | WIMPy | elliot98: Registrations hav absolutely no relation to outgoing calls (or vice versa). |
| 17:33.39 | WIMPy | mathi: Why do you think so? |
| 17:34.31 | mathi | WIMPy, powersuply generator 1000$ - $3000, battery set that can suply minimum 3 hours standalone work - 2000$ - 3000$, minimum 2 or 3 internet provider connections for approve nice bandwidth 24/7, plus the server itself, ... |
| 17:34.32 | elliot98 | WIMPy: registration let's a peer send calls in even though it's not marked as a user? |
| 17:35.31 | WIMPy | elliot98: The only function of registering is to tell the other end where to reach you. |
| 17:35.58 | WIMPy | mathi: Use a notebook. That has built-in batteries for some hours. |
| 17:37.01 | mathi | are you serious?) is that powerful enough to handle 4 simultaneous calls? |
| 17:37.05 | elliot98 | WIMPy: so I would still need to mark a provider/peer also as a user if I want asterisk to accept calls from that provider |
| 17:37.21 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 17:37.57 | r0m|u | mathi, I have an embedded system with 500MHz CPU and 256MB of RAM and handles 6 simultaneous calls just fine. imagine what a nice laptop can do |
| 17:38.08 | WIMPy | mathi: I wouldn't see why a small atom netbook couldn't do it. If it wasn;t for RAM size you could prbably do it on a router with OpenWRT or something. |
| 17:38.23 | mathi | nice:) |
| 17:38.36 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 17:38.56 | r0m|u | There is a math to all this. base on con current calls/bandwidth/cpu/ram |
| 17:39.22 | WIMPy | elliot98: Users have to authenticate to you. ITSPs usually won't so you define a peer for them. |
| 17:40.03 | *** join/#asterisk cerberus_za (~coert@8ta-151-10-40.telkomadsl.co.za) |
| 17:40.36 | elliot98 | WIMPy: but does asterisk accept calls coming from peers, because officially asterisk should only accept calls from useres |
| 17:40.39 | elliot98 | *users |
| 17:40.41 | r0m|u | 64kbpsX2X2chanXconcurrentcalls=bandwidth requirements |
| 17:41.23 | WIMPy | elliot98: What makes you think so? |
| 17:41.27 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 17:41.37 | WIMPy | The difference is how users and peers are matched. |
| 17:41.52 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 17:42.21 | elliot98 | WIMPy: I had a client set up as a peer and it wasn't working until I set it as a user, it was constantly giving a "not authorized" on an invite |
| 17:42.30 | [TK]D-Fender | r0m|u, 85kbps <- |
| 17:42.42 | r0m|u | o ok. Thanks [TK]D-Fender |
| 17:43.06 | [TK]D-Fender | elliot98, depends how they sent you auth, if at all |
| 17:43.26 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 17:43.40 | elliot98 | it was a voip client, so once I changed it to "user", things worked |
| 17:44.06 | [TK]D-Fender | elliot98, Peer usually coveres everything unless you've done something wrong. |
| 17:44.20 | WIMPy | See, there's the differenc of you registering to an ITSP or a user registering to you. |
| 17:44.47 | elliot98 | so the username/secret in account for a peer would be used on outgoing calls? |
| 17:45.10 | elliot98 | whereas the username/secret for users would be used on incoming calls? |
| 17:45.39 | WIMPy | yes; and no, both ways. |
| 17:45.40 | hardwire | r0m|u: radio shack! |
| 17:45.51 | WIMPy | In that order |
| 17:46.21 | [TK]D-Fender | elliot98, peer matches on IP first. User matches on name |
| 17:46.21 | r0m|u | hardwire, I did. when I mention 3.3 they looked at me like I was crazy... lol any how I am using a regular 3v.... thats why I ask |
| 17:46.41 | hardwire | most radio shack employees look at people crazy |
| 17:46.45 | r0m|u | lol |
| 17:46.49 | elliot98 | but other then that users and peers are essentially the same |
| 17:46.50 | r0m|u | so true |
| 17:46.54 | hardwire | go ask for a DVI cable.. then say "I don't need one that's gold plated" and then.. behold.. |
| 17:46.57 | hardwire | crazy face |
| 17:47.01 | r0m|u | rofl |
| 17:47.15 | hardwire | r0m|u: cr2032 is all you need |
| 17:47.22 | hardwire | bbl |
| 17:47.22 | r0m|u | so true |
| 17:47.33 | r0m|u | I figured. Got one. all working! |
| 17:47.35 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ilvcviyruvyxdnek) |
| 17:47.37 | r0m|u | Thanks for the pointers |
| 17:47.39 | hardwire | yup |
| 17:47.42 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ilvcviyruvyxdnek) |
| 17:49.03 | *** join/#asterisk garymc (~chatzilla@host81-139-152-192.in-addr.btopenworld.com) |
| 17:50.36 | elliot98 | and it nothing matches, asterisk will ask to authenticate using whatever username/secret is given in the account? |
| 17:50.37 | *** join/#asterisk bluregard (~matt@c-98-228-3-34.hsd1.il.comcast.net) |
| 17:50.50 | bluregard | hi all |
| 17:51.10 | mathi | WIMPy, imagine of the laptop breaks, asterisk is down; can I just put the card with the lines into another computer and start running asterisk on that computer? |
| 17:51.13 | elliot98 | *if |
| 17:51.15 | p3nguin | type=user only allows calls into asterisk; calls cannot go outbound from asterisk if the type is user. |
| 17:51.35 | elliot98 | p3nguin: and peer can do both? |
| 17:51.49 | p3nguin | type=user matches on username |
| 17:51.51 | p3nguin | type=peer matches on IP/port, and allows calling in both directions. |
| 17:51.53 | bluregard | anyone around that can speak to the usage of AMI? |
| 17:52.11 | p3nguin | type=friend is a hybrid of type user and type peer, and it allows calling in both directions. |
| 17:52.23 | elliot98 | p3nguin: but if there is no peer with IP/port, then can it authenicate with username/secret |
| 17:52.30 | WIMPy | mathi: On a Laptop you'd probably have to go for USB anyway. So that's easy to plug ion to the next laptop. |
| 17:52.50 | elliot98 | or will it be necessary to set the account as a "friend" |
| 17:53.00 | p3nguin | As far as I know, peer does not ever perform matching on username. It will use the username to authenticate, though. |
| 17:53.19 | elliot98 | p3nguin: ok, so that would explain why I needed to set the account as "user" , not "peer" |
| 17:53.22 | WIMPy | bluregard: Ask you question and see what happens. Metaquestions are usually ignored. |
| 17:53.42 | p3nguin | If you want to match on username and have calls work in both directions, use type=friend. |
| 17:53.44 | mathi | WIMPy, is it that easy to recover from a hardware failure? |
| 17:53.59 | bluregard | yeah that usually doesn't work out too well for me but why not. |
| 17:54.20 | WIMPy | mathi: If you've got a 2nd sytem that is configured the same. |
| 17:54.34 | r0m|u | mathi, get the next laptop plug it in to the usb and off you go. |
| 17:54.38 | elliot98 | now, if insecure=port,invite set on a friend, would the call go through without any auth if the username is set? |
| 17:54.46 | bluregard | first of all I'm trying to find documentation on all of the various Event:'s in 1.8 and am finding that a bunch are undocumented. |
| 17:54.46 | r0m|u | with same specs |
| 17:54.48 | p3nguin | no |
| 17:54.56 | mathi | r0m|u, which specs ? |
| 17:55.22 | r0m|u | mathi, if the laptop brakes have one with the same hardware config |
| 17:55.26 | elliot98 | p3nguin: ok, because that'll be a security issue |
| 17:55.28 | r0m|u | "specs" |
| 17:55.39 | bluregard | second, I'd like to know if there's any way to associate an Event: with the Action: that initiated it. I thought the ActionID: might work, but it doesn't look like it. |
| 17:55.40 | mathi | r0m|u, why is that necessary? |
| 17:55.48 | mathi | r0m|u, it could be two different laptops |
| 17:56.10 | p3nguin | elliot98: http://www.voip-info.org/wiki/view/Asterisk+sip+insecure |
| 17:56.20 | WIMPy | bluregard: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Manager+Interface+(AMI) |
| 17:56.47 | r0m|u | mathi, if one brakes and it was running of the usb and you need to boot of the new system and the hardware specs are not the same you can run in to issues specially if the architecture does not match |
| 17:56.49 | bluregard | yes, that is the first place I went for documentation |
| 17:56.50 | *** join/#asterisk osas (~osas@nslu2-linux/osas) |
| 17:56.59 | WIMPy | bluregard: And the action ID is only for the response to a command. If you want to track the call further, you need to check the uniqueid. |
| 17:57.28 | r0m|u | mathi, linux is good about recognizing hardware but is not THAT good. |
| 17:57.42 | elliot98 | p3nguin: the page is a bit confusing, because insecure=port implies that we're doing IP authentication, and insecure=invite implies we are doing username/secret auth |
| 17:57.42 | bluregard | WIMPy: right, but the uniqueid isn't conveyed in the Response: that I can tell |
| 17:57.50 | osas | srry if this was posted, but http://www.asterisk.org/ is down: The mysqli error was: Too many connections. |
| 17:58.04 | mathi | r0m|u, I'll first make sure that the backup server can run asterisk properly |
| 17:58.13 | r0m|u | mathi, modules could also create issues. |
| 17:58.14 | mathi | in that case it doesn't matter |
| 17:58.23 | WIMPy | r0m|u: Are you thinking about taking th HDD from one system to another? |
| 17:58.25 | bluregard | response just tells me that the action was queued or not, not that asterisk actually did anything with it. |
| 17:58.37 | p3nguin | elliot98: I'm sorry you interpret it that way. |
| 17:58.53 | r0m|u | WIMPy, if one system pukes and he is running of an external usb than yes. |
| 17:59.03 | WIMPy | bluregard: What exactely are you doing? You usually get an acknowledge resopnse and a completed response. |
| 17:59.31 | bluregard | WIMPy: originate, lots and lots of originates |
| 17:59.34 | WIMPy | r0m|u: I was just talking about USB for the line interfaces. |
| 17:59.39 | elliot98 | p3nguin: how is it to be understood? |
| 18:00.02 | mathi | WIMPy, actually I could run linux and asterisk on external HDD in RAID, if one laptop crash, I only need to connect the card with lines + the HDD |
| 18:00.03 | r0m|u | WIMPy, I see. mathi in that case ignore what I said. |
| 18:00.08 | p3nguin | elliot98: type=peer matches on IP/port. Using insecure=port causes the port to be ignored. |
| 18:00.18 | r0m|u | mathi, bad idea! |
| 18:00.22 | mathi | r0m|u, why ? |
| 18:00.25 | r0m|u | raid on usb = disaster waiting to happen |
| 18:00.25 | p3nguin | elliot98: Leaving only the IP to match. |
| 18:00.32 | mathi | oh ... |
| 18:00.41 | mathi | sounds you experienced that? :) |
| 18:00.47 | elliot98 | p3nguin: what does insecure=invite do? |
| 18:01.12 | r0m|u | you could use label on the fs to circumvent some of the problems but still a bad idea |
| 18:01.20 | WIMPy | Yes, you have to take a little care with USB-storage. If it is used by something with a higher priority than itself, it can deadlock. |
| 18:01.30 | r0m|u | ^^ |
| 18:01.31 | p3nguin | elliot98: insecure=invite makes it so that the peer does not have to authenticate a second time (during the INVITE); once it has authenticated the first time, INVITES are then trusted. |
| 18:02.15 | p3nguin | elliot98: When you don't use insecure=invite, you can see in a sip debug where an INVITE will return an unauthorized message, causing the peer to have to authenticate again. insecure=invite remove that. |
| 18:02.59 | *** join/#asterisk cerberus_za (~coert@8ta-151-5-216.telkomadsl.co.za) |
| 18:03.06 | p3nguin | elliot98: If you have no problem authenticating with the INVITE, don't use insecure=invite. |
| 18:03.13 | elliot98 | p3nguin: but peers doesn't use username/secret, so how could it authenticate? |
| 18:03.59 | p3nguin | elliot98: They do *use* username and secret, but they don't *match* on username. |
| 18:04.20 | p3nguin | elliot98: The match is done by IP address and port for type=peer. |
| 18:04.48 | mathi | WIMPy, how can i connect a card (you are talking about these TDM cards?) through USB ? |
| 18:05.10 | p3nguin | elliot98: If you need it to match the peer entry by the username, use type=friend. |
| 18:05.17 | elliot98 | p3nguin: so I would need to set the proper IP address for the account and only then would it check the username/secret in that account |
| 18:05.22 | WIMPy | mathi: Use an USB one. |
| 18:05.28 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 18:05.38 | mathi | WIMPy, can I you give a model as an example ? I will look it up now |
| 18:05.42 | mathi | *can you |
| 18:06.00 | elliot98 | p3nguin: unless I set insecure=invite, so it will only look at the IP address |
| 18:06.17 | elliot98 | p3nguin: and won't ask for any more credentials |
| 18:06.20 | p3nguin | elliot98: insecure=invite makes it not authenticate during the invite. |
| 18:06.35 | kaldemar | peers are matched by username also. |
| 18:06.42 | elliot98 | p3nguin: so if the IP matches, it will let the call through |
| 18:07.06 | p3nguin | Peers are matched by IP/port, not username. |
| 18:07.12 | p3nguin | type=peer, that is. |
| 18:07.13 | bluregard | WIMPy: I'm seeing the ack response but not any kind of completed response. |
| 18:07.22 | elliot98 | p3nguin: now what if insecure=invite is set for a user, what will happen? |
| 18:07.25 | WIMPy | mathi: X-Tensions XC-525 or Trust 13018, but similar ones exist from many vendors. |
| 18:08.09 | p3nguin | elliot98: The user (type=user) will match the peer entry by the username, and it will not be required to authenticate during an INVITE. |
| 18:08.13 | *** join/#asterisk DennisG (~dennisg@ip5454b5b3.adsl-surfen.hetnet.nl) |
| 18:08.15 | kaldemar | p3nguin: see "naming devices" in the sample config. |
| 18:08.40 | elliot98 | p3nguin: so if insecure=invite, anyone can send a call with a username and the call will go through |
| 18:08.54 | elliot98 | if it's a user account |
| 18:08.58 | WIMPy | bluregard: It's been some time since I've tried that, but I somehow managed to get soem relation. Are you using asynchronous? |
| 18:09.33 | elliot98 | p3nguin: since it matched with username, insecure=invite will not require any password |
| 18:09.36 | bluregard | WIMPy: more than likely. I'll need to do this in a non-blocking fasion |
| 18:10.01 | elliot98 | p3nguin: this is ok for peers that authenticate with IP addresses, but for users, any device in the "cloud" can send calls |
| 18:10.11 | WIMPy | bluregard: Maybe that's the difference. |
| 18:10.16 | elliot98 | p3nguin: as long as they know the username of a device |
| 18:10.19 | p3nguin | kaldemar: As far as I can tell, numbers 1, 2, and 3 say exactly what I said. |
| 18:10.49 | p3nguin | type=user matches on username; type=peer matches on IP/port. |
| 18:11.24 | mathi | WIMPy, you also suggested me yesterday to use BRI? what is the difference with these cards |
| 18:11.43 | WIMPy | Those adapters are for BRIs. |
| 18:11.52 | bluregard | WIMPy: what if I passed a channel var along with the originate and use that to find the uniqueid? Or is that more convoluted than it needs to be? |
| 18:12.09 | p3nguin | elliot98: I see what you're saying, but I've never investigated that theory. I'm sure that's not the case, though. |
| 18:12.15 | WIMPy | bluregard: That should work. |
| 18:12.24 | p3nguin | Having a setting to open it up for anyone knowing the username seems bad. |
| 18:12.52 | elliot98 | p3nguin: yes, but theoretically, that is how things would come out |
| 18:13.10 | Micc | elliot98, I've had the same worry, but have not seen it in practice. |
| 18:13.10 | mathi | WIMPy, i only read in the description that it is a "usb isdn adapter" |
| 18:13.24 | p3nguin | In theory, I see your concern. In reality, there is probably something else that I don't know about which keeps that from happening. |
| 18:13.42 | WIMPy | mathi: Yes. |
| 18:13.56 | mathi | owkay .. wikpedia BRI :) |
| 18:14.00 | *** join/#asterisk mnicholson (~mnicholso@nat/digium/x-vzjwolrlszpcycdo) |
| 18:14.00 | *** mode/#asterisk [+o mnicholson] by ChanServ |
| 18:14.01 | Micc | elliot98, I've tested a phone without registering and using invalid/username and password, but from an IP that had other registered phones, so it let it through. |
| 18:14.02 | WIMPy | ISDN comes in two sized: BRI or PRI |
| 18:14.28 | Micc | elliot98, which is a little scary in itself, but at least I don't think it would allow it to work from any IP that has not had any devices registered. |
| 18:14.35 | mathi | WIMPy, and E1, T1, J1 comes in which category ? |
| 18:14.41 | p3nguin | Basically, using insecure=invite is a big risk. |
| 18:14.56 | WIMPy | mathi: That's lines carrying PRIs. |
| 18:15.11 | r0m|u | p3nguin, does this apply to itsp? |
| 18:15.24 | p3nguin | micc: But change to type=user and then send a call from a device which never registered, but using the valid username. |
| 18:15.25 | r0m|u | so we are insecure regardless? |
| 18:15.39 | mathi | WIMPy, what is the differnece between a BRI and a PRI? |
| 18:15.56 | Qwell | about N |
| 18:16.11 | WIMPy | mathi: Size. BRI=2 channels, PRI=30 channels (or 24). |
| 18:16.42 | mathi | WIMPy, we were talking about a 4 channel capacity, that would imply that you suggest me to purchase two of these adapters ? |
| 18:16.55 | Micc | p3nguin, all my users are peers because I read somewhere a while back that the difference between them was going away so it didn't matter. |
| 18:17.00 | p3nguin | r0m|u: If someone else would send a call from the ITSP's IP address, the call could make it through. |
| 18:17.02 | WIMPy | mathi: yes. |
| 18:17.15 | Micc | p3nguin, I would need to do a lot of testing to be able to switch all my customers to users or friends instead of peers. |
| 18:17.37 | r0m|u | p3nguin, ip spoofing is not hard at all. I should start looking in to this. didnt know. Thank you very much for the info p3nguin |
| 18:17.46 | *** part/#asterisk libryder (~david@209.33.214.243) |
| 18:17.49 | p3nguin | micc: Well, type=user only allows calls to go from the device to asterisk; asterisk cannot send calls to a peer which is configured as type=user... |
| 18:17.54 | WIMPy | mathi: You could even take 4 and wire up both your active and your backup system. |
| 18:18.24 | Micc | p3nguin, so I would need to at least be friend, but would that solve the security problem? |
| 18:18.45 | mathi | WIMPy, you mean two in each? or really plug 4 of those to both of my servers? |
| 18:18.51 | p3nguin | micc: But type=friend should also match on username like user does, but still retain much of the type=peer characteristics as well. |
| 18:19.02 | WIMPy | mathi: two each |
| 18:19.12 | mathi | yes that's a smart idea :) |
| 18:19.33 | mathi | but then I need to synchronize the MySQL DB's |
| 18:19.48 | Qwell | osas: Are you still having issues now? |
| 18:19.53 | Micc | can insecure be set on individual peers or only in general? |
| 18:20.03 | Micc | I have only 1 or 2 customers that need insecure=invite |
| 18:20.21 | osas | Qwell: works ok now |
| 18:20.40 | p3nguin | It should only be used per peer. |
| 18:20.44 | Qwell | osas: thanks. we were having issues internally. I wanted to make sure that clearing that up fixed your issue |
| 18:20.53 | osas | sure, np |
| 18:21.03 | WIMPy | mathi: If your telco supports bundling of ptmp lines, you could even have both systems active simultaneousely. |
| 18:21.27 | WIMPy | But then that may not be a good idea with whatever you run un them. |
| 18:21.39 | WIMPy | i.e. your application. |
| 18:22.34 | Micc | p3nguin, actually, now that I look all my users are friends. That makes me feel a little better. |
| 18:22.40 | elliot98 | can users register at all? I'm getting No matching peer found when a type=user tries to register |
| 18:23.05 | p3nguin | Peers set as type=user can register if the host is set to dynamic. |
| 18:23.16 | mathi | WIMPy, why wouldn't I be able to plug two in one server, and two in another server, and get calls simultaneously on both servers ? |
| 18:24.54 | elliot98 | p3nguin: host is set to dynamic |
| 18:26.29 | p3nguin | Make sure there is a defaultuser and a secret -- the peer will try to match based on that. |
| 18:26.51 | elliot98 | p3nguin: defaultuser, not username? |
| 18:27.00 | p3nguin | depends on the asterisk version |
| 18:27.02 | *** part/#asterisk cVsup (~cVsup@201.78.47.2) |
| 18:27.19 | p3nguin | If it's old, username; if it's new, defaultuser. |
| 18:27.49 | p3nguin | (but username should still work on a new system, showing a warning about the change) |
| 18:28.24 | elliot98 | ok |
| 18:29.26 | elliot98 | p3nguin: still failed, seems users don't register |
| 18:29.54 | p3nguin | They should be able to... when there isn't some other unknown problem. |
| 18:30.12 | p3nguin | I'll test. |
| 18:31.10 | elliot98 | p3nguin: this is on version 1.4 |
| 18:32.07 | elliot98 | p3nguin: thanks |
| 18:33.25 | Katty | so. much. groggy. still. |
| 18:33.39 | Katty | somehow i got too much sleep |
| 18:34.41 | bluregard | WIMPy: it is the async: yes that provides the OriginateResponse which contains both the actionID and uniqueID. Thank you. |
| 18:35.42 | p3nguin | elliot98: I am able to duplicate your issue. |
| 18:36.02 | p3nguin | elliot98: Now I just have to see if I can understand what prevents the match and the registration. |
| 18:36.27 | elliot98 | p3nguin: interesting...now what would happen if you send a call with the username on an account with type=user and insecure=invite |
| 18:36.56 | p3nguin | elliot98: I'll test that, too. |
| 18:37.09 | Micc | Its a little scary all the ways you can make an asterisk server insecure if you don't know how it works. I feel even after years of working with asterisk I still don't know how some things work. |
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| 18:38.21 | elliot98 | p3nguin: thanks |
| 18:38.21 | p3nguin | elliot98: Okay, you want to know what happens? :) |
| 18:38.37 | elliot98 | p3nguin: drumroll rat-tat-tat |
| 18:38.38 | p3nguin | elliot98: It allows the call, even with the wrong password. |
| 18:38.58 | elliot98 | so basically type=friend should never ever use insecure=invite |
| 18:39.03 | elliot98 | p3nguin: this is obsurd |
| 18:39.14 | hacim | can anyone recommend a good termination provider? |
| 18:39.38 | elliot98 | p3nguin: think this should be brought to someone's attention?? |
| 18:39.47 | Micc | elliot98, I assume that would apply to type=peer as well. |
| 18:40.10 | Micc | p3nguin, do you have other phones that have registered from that same network? |
| 18:40.20 | Micc | this freaked me out too when I tested it. |
| 18:40.20 | p3nguin | network yes, same IP address no |
| 18:40.31 | elliot98 | Micc: unless peers don't match all with username |
| 18:40.32 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 18:40.53 | p3nguin | type=friend behaves in the same manner... I was able to send a call without being registered and without the correct password. |
| 18:41.21 | elliot98 | Micc: so although IP a spoofing can done, it's not nearly as vulnerable as when type=user |
| 18:42.00 | p3nguin | I use type=peer in nearly every entry, so I never noticed how the insecure setting opens that up. |
| 18:43.43 | p3nguin | My test could also be flawed, since I was testing from a device that had once been registered already. |
| 18:43.46 | Micc | p3nguin, so your saying it wasn't just because I had other registered devices from that IP? |
| 18:44.04 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 18:44.06 | p3nguin | If you are using type=peer and insecure=invite, it probably was because of that. |
| 18:44.07 | Micc | p3nguin, ok lets get a valid test here, your freaking me out. |
| 18:44.57 | Micc | I should have went a step further to verify this when I noticed it, but I assumed it was fine since I get hacking attempts all the time, but none seem to be able to make a call. |
| 18:45.13 | p3nguin | Since type=peer matches on IP/port, the username is irrelevant for peer matching. Authentication during invite is something else, though, and removing the necessity to authenticate in the invite probably allows other devices from the same IP address to make calls. |
| 18:45.18 | Micc | So I would assume that its fine. Unless all they would need to know is the correct username, that would be scary. |
| 18:45.59 | Micc | p3nguin, if they have to be from the same IP, I'm ok with that. |
| 18:46.21 | p3nguin | I think ideally you will use type=peer and insecure=no. |
| 18:46.49 | p3nguin | That will match the peer by the IP address and force the device to auth in each invite. |
| 18:46.53 | p3nguin | (as far as I can tell) |
| 18:46.57 | Micc | I'm testing on one of my servers with insecure=no and the one customer that needs it has insecure=invite. |
| 18:47.47 | Micc | There are some routers that have a problem with the invite needing to be authenticated. |
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| 18:48.46 | Katty | hi leif |
| 18:48.51 | leifmadsen | ophai |
| 18:49.14 | eppigy | nein |
| 18:49.15 | eppigy | NEIN |
| 18:49.18 | WIMPy | mathi: Yes, you can do that on ptmp lines. But you should check with your telco if they provide bundles of ptmp lines with the same number(s). |
| 18:49.41 | Katty | hi eppigy |
| 18:49.47 | eppigy | yellow |
| 18:50.48 | elliot98 | Micc: well, also providers don't usually authenticate at all |
| 18:51.02 | mathi | WIMPy, i'm still very new to all of this, basically I would have a single number, and clients would transfer/forward to that number? |
| 18:52.25 | WIMPy | mathi: Yes. Or you can get DDI with many numbers. You might be able to extends them to get magnitudes more, but I don't know how they configure that in Belgium. |
| 18:54.07 | mathi | WIMPy, why would I need that? |
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| 18:54.58 | Micc | elliot98, yeah but host != dynamic for providers. I only put in the IPs that they tell me. |
| 18:55.04 | WIMPy | mathi: In case you get more business ideas or if some of your customers can't forward calls in a way that will let you see both their number and the callers number. |
| 18:55.39 | mathi | WIMPy, and a forward is free ? |
| 18:55.47 | Micc | elliot98, so I suppose if you were looking for a way to make some bad calls you could spoof a provider's IP, but the acks and such wouldn't make it back and forth, so it would be really difficult even then. |
| 18:56.03 | WIMPy | mathi: Probably not. |
| 18:56.07 | elliot98 | Micc: yes, so it is kind of useless for a real call |
| 18:56.36 | elliot98 | Micc: but if peers match on username too, that would be an issue |
| 18:56.39 | mathi | WIMPy, does the client need to pay this or is it my responsability? |
| 18:56.45 | Micc | spoofing a single UDP packet is easy. Getting the timing and the tokens right to actually make a call would be almost impossible without some sort of sniffing too. |
| 18:56.47 | elliot98 | so peers should only be peers, not friends |
| 18:57.15 | WIMPy | mathi: The forwarding is dome by the doc, so it depends on his play if or how much he pays for that. |
| 18:57.18 | elliot98 | because then it may match on username and if insecure=invite, then it's asking for trouble |
| 18:57.20 | Micc | yeah, for providers I use peers. |
| 18:57.58 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v015-242.mobile.uci.edu) |
| 18:57.59 | Micc | elliot98, right, if host=dynamic too. |
| 18:58.19 | WIMPy | mathi: And it may also depend on if you're using the same telco as him. |
| 18:59.06 | WIMPy | Unconditional on-net transfers are free with some telcos. |
| 18:59.14 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 18:59.25 | mathi | WIMPy, what is on-net ? |
| 18:59.39 | WIMPy | The same network. |
| 19:00.41 | mathi | I see. Now all I need to know is if I will receive the caller number + doctor number. Before I invest in anything... |
| 19:01.37 | WIMPy | If the doc forwards via the telcos switch that will work. If he does it in his PBX it might fail. |
| 19:02.14 | mathi | WIMPy, there is no workaround in the latter case ? |
| 19:02.36 | mathi | (if it fails) |
| 19:02.55 | WIMPy | The workaround would be to find out how to tell teh PBX not to do it itself, but to tell the switch to do it. |
| 19:03.08 | Micc | elliot98, I would still like to know how long asterisk holds the IP address in memory that it uses for the insecure=invite. I would think its whatever the registration expiry time is. |
| 19:03.51 | mathi | WIMPy, in the case of the switch, we talk about a forward. In the case of the PBX, a transfer. Right? |
| 19:04.13 | WIMPy | No, forwarding in both cases. |
| 19:05.51 | mathi | WIMPy, isn't it possible to change the caller number ? |
| 19:06.00 | mathi | (PBX side) |
| 19:06.02 | WIMPy | Where? |
| 19:06.08 | mathi | ^ |
| 19:06.23 | *** join/#asterisk hudony (~chatzilla@modemcable078.217-70-69.static.videotron.ca) |
| 19:06.44 | WIMPy | In theory, yes. If there's configuration option to do so is another matter. |
| 19:07.01 | WIMPy | Unfortunately there are amny ways PBXes can work. |
| 19:07.06 | hudony | hi : simple question : if my asterisk server is installed on a linux box ascting as a gateway, do I have to cinfigure asterisk as nat with externip nat=yes etc? |
| 19:07.28 | hudony | Thx box has 2 interfaces... so im confused |
| 19:07.32 | WIMPy | hudony: no |
| 19:07.42 | [TK]D-Fender | hudony, .. |
| 19:07.44 | [TK]D-Fender | ~sipnat |
| 19:07.45 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 19:07.46 | [TK]D-Fender | ^^ |
| 19:07.53 | [TK]D-Fender | hudony, and no. |
| 19:07.56 | WIMPy | It is reachable with it's own IP. |
| 19:08.11 | hudony | yes of course via the public interface |
| 19:08.13 | hudony | ok thank you |
| 19:08.29 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v015-242.mobile.uci.edu) |
| 19:08.34 | hudony | I was asking it to be sure cause my problem is when someone calls in, I cannot hear him but he can hear me |
| 19:08.49 | hudony | Just like the rtp packets were not reaching the phone |
| 19:08.51 | hudony | :S |
| 19:09.07 | mathi | WIMPy, so i'll have to ask the user to enter their phone number probably if doc has pbx( |
| 19:09.21 | WIMPy | hudony: And the wirewall will let them in? |
| 19:09.35 | hudony | actually...firewall is disabled for testing purpose |
| 19:09.41 | hudony | so it shouldnt be an issue |
| 19:10.27 | hudony | phone status gives me : 2067 rtp pacekts sentand 0 packets received |
| 19:10.37 | WIMPy | mathi: Better to find out how to configure it in that case. Or set up the forwarding via other customer service means (telcos hotline or webinterface). |
| 19:10.38 | hudony | so i guess the problem is really related to that |
| 19:11.35 | WIMPy | I used to set canreinvite=no for phones on the LAN, i.e. behind NAT. But I think I have enabled everything now and it still works. |
| 19:11.53 | mathi | WIMPy, you mean how to configure the PBX to not treat the call and forward it instead? Or something else? |
| 19:12.12 | mathi | (^forward with switch instead) |
| 19:12.25 | WIMPy | Yes, or do it the way you need it. |
| 19:12.45 | *** join/#asterisk libryder (~david@209.33.214.243) |
| 19:13.13 | mathi | or do how |
| 19:13.20 | mathi | didn't get you |
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| 19:13.54 | WIMPy | That's where a bunch of numbers can be handy, but if it fails you likely get the forwarder as caller so it's more likely for the callerID to get lost in a PBX scenario. |
| 19:15.11 | WIMPy | OTOH they might call with withheld number or from another persons phone anyway. So you need to be able to identify them in another way anyway. |
| 19:15.18 | mathi | well I guess if I have a bunch of numbers it won't solve anyhow the PBX scenario, so that would be useless |
| 19:15.42 | libryder | grr I can't figure out why a module I have installed isn't getting loaded... |
| 19:15.46 | libryder | [Nov 7 14:14:22] WARNING[31139] res_agi.c: Could not find application (Swift) |
| 19:16.10 | mathi | WIMPy, yes sure, and from phpne office etc. I plan to confirm their number, if not right one, only then they have to type in their phone number |
| 19:16.15 | libryder | I have app_swift.so in /usr/lib/asterisk/modules and swift.conf in /etc/asterisk/ |
| 19:16.20 | mathi | *phone in office |
| 19:16.41 | libryder | and I've restarted asterisk |
| 19:17.21 | libryder | swift is installed and [swift -o test.mpt "hello world"] completes successfully |
| 19:17.26 | WIMPy | mathi: Indeed that's not the part that is likely to be the issue. It will be the callers. |
| 19:17.28 | libryder | mp3* |
| 19:17.36 | [TK]D-Fender | hudony, reinvites still need to be disabled |
| 19:18.12 | [TK]D-Fender | hudony, And it doesn't mean that you might not have to set nat=yes for your peers |
| 19:18.12 | mathi | WIMPy, but still I don't see how a bunch of numbers might anyhow help me. The caller ID get lost from the PBX |
| 19:18.29 | hudony | oh |
| 19:18.31 | hudony | :S |
| 19:18.32 | WIMPy | That's what I said. |
| 19:18.33 | hudony | ok |
| 19:18.38 | elliot98 | Micc: I am a bit confused, because even after unregistering, calls seem to still associate with the host=dynamic peer |
| 19:18.43 | WIMPy | It's not likely the issue. |
| 19:19.20 | mathi | WIMPy, though you suggested that system earlier as an option |
| 19:19.26 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 19:19.40 | SuperNull | anyway to for certain disable MWI Notification ? |
| 19:20.04 | [TK]D-Fender | SuperNull, don't put their box in their peer |
| 19:20.31 | SuperNull | we use an external.. MWI APP we made.. and it worked fine till 1.8 .. now 1.8 sends notifys as well and cancels my remote MWI Notify being sent |
| 19:20.41 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 19:20.44 | WIMPy | Yes, it can't hurt and will definitely be available over a bundle of lines. The drawback is that you probably cannot have multiple active devices per line. |
| 19:21.46 | [TK]D-Fender | SuperNull, Only sends it if you specified the mailbox |
| 19:21.56 | SuperNull | hurmmm. |
| 19:22.47 | SuperNull | well thats a problem, we use realtime config for that.. |
| 19:22.57 | SuperNull | if i kill the vm module in this instance will that do any good for this ? |
| 19:24.00 | [TK]D-Fender | remove the box entry |
| 19:27.32 | elliot98 | Micc: it seems that insecure=invite is a peer only option |
| 19:27.41 | elliot98 | Micc: users always need to authenticate |
| 19:27.57 | elliot98 | Micc: this is what I see from some initial tests |
| 19:28.09 | elliot98 | Micc: but more study needs to be done as to what is happening |
| 19:31.48 | SuperNull | Well [TK] that sucks. |
| 19:32.03 | SuperNull | i removed it from our pbx_user table as requested and that made it stop bitching. |
| 19:32.09 | libryder | is there a way to find out asterisk's modules path? |
| 19:33.04 | leifmadsen | look in asterisk.org |
| 19:33.09 | leifmadsen | s/asterisk.org/asterisk/conf/ |
| 19:33.11 | leifmadsen | back |
| 19:33.13 | leifmadsen | bah! |
| 19:33.19 | leifmadsen | look in asterisk.conf |
| 19:33.33 | leifmadsen | or 'core show settings' |
| 19:33.43 | libryder | thanks! |
| 19:36.46 | mathi | WIMPy, the slots in the cards are for RJ-45 cables? |
| 19:38.37 | WIMPy | "RJ-45", yes. |
| 19:39.36 | mathi | WIMPy, I don't have a working slot already then? |
| 19:39.47 | WIMPy | ??? |
| 19:40.15 | mathi | WIMPy, well I have a slot for an rj-45 cable, I use it for internet now ... |
| 19:40.26 | mathi | the card doesn't do isdn i guess |
| 19:40.30 | mathi | :) |
| 19:40.39 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 19:40.40 | WIMPy | That's ethernet. |
| 19:41.08 | WIMPy | There are many interfaces using 8P8C modular connectors. |
| 19:42.38 | elliot98 | p3nguin Micc : from what I see, if a secret is not set for a user, it matches on the user without any need for authentication |
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| 19:55.57 | blizzow | Are there any incompatibilities between the latest DAHDI and a red-fone? |
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| 20:00.20 | [TK]D-Fender | blizzow, I believe there have been recent issues with TDMoE over the past 2 months from redfone users. rolling back a version seemed to clear things up |
| 20:04.25 | blizzow | Thanks. |
| 20:14.42 | Micc | elliot98, I assume someone has thought about these issues long and hard and has implimented it in the correct way. But I think we really need some clarification from that person as to what is really going on there. |
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| 20:27.58 | jeffspeff | can you set the [general] context of sip.conf to be read from a realtime mysql? |
| 20:28.26 | wdoekes2 | static realtime |
| 20:28.57 | wdoekes2 | i.e. the config file as is, but from a database instead of from the filesystem |
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| 20:30.10 | jeffspeff | yes, i know how to set realtime peers and users, however whatabout the other fields within [general]? just add them to the same db but give them null values for everything except for the general context? |
| 20:30.52 | wdoekes2 | you can leave out any columns you like |
| 20:31.02 | *** join/#asterisk garymc (~chatzilla@host86-176-88-100.range86-176.btcentralplus.com) |
| 20:31.08 | wdoekes2 | if you're not overriding the setting for any realtime peer, you don't need a column for it |
| 20:41.02 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 20:45.06 | elliot98 | is there any reason to have both realtime peers and users? Wouldn't peers be sufficient? |
| 20:47.42 | r0m|u | Is it ok to run several sip phones on the same port ie 5060? |
| 20:48.06 | r0m|u | or should I change each one of them to a different port? |
| 20:48.13 | wdoekes2 | elliot98: if you want people to call without having registered first, you need users |
| 20:48.40 | wdoekes2 | r0m|u: you mean behind nat? |
| 20:49.35 | r0m|u | wdoekes2, all internal inside my network talking to my asterisk server |
| 20:50.10 | wdoekes2 | if your asterisk is internal too, then there's no need to switch ports |
| 20:51.52 | r0m|u | Thank you. thats what I thought. |
| 20:54.21 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 20:56.03 | elliot98 | wdoekes2: I see, but it seems most voip clients register, so it usually should not be an issue |
| 20:57.49 | [TK]D-Fender | <wdoekes2> elliot98: if you want people to call without having registered first, you need users <- incorrect |
| 20:58.31 | [TK]D-Fender | You do not need to be registred to place calls |
| 20:58.32 | elliot98 | [TK]D-Fender: so what's correct? |
| 20:59.22 | [TK]D-Fender | elliot98, the difference between peer & user is on name vs IP ordering for matchin, and "user" is normally only required for providers, not phones. |
| 20:59.31 | Naikrovek | that whole "you don't need to register to make calls" seems like it should be configurable. I don't *ever* want anyone to make a call on my system without being registered. |
| 21:00.10 | *** join/#asterisk LittleFool (~LittleFoo@95.129.212.120) |
| 21:00.29 | [TK]D-Fender | elliot98, Which is why people also run into issues of digest user issues with multiple accounts on a single phone when using type=peer, because the wrong one gets matched first. Switch to 'friend (which is a combined user/peer) and that problem goes away |
| 21:00.30 | elliot98 | [TK]D-Fender: p3nguin before ran a test and set an account as a peer and was not able to authenticate an INVITE |
| 21:00.55 | Naikrovek | i know you can do it with contexts and stuff, prevent calls from unregistered peers, but ... shouldn't that be the default? I dunno. contexts still confuse me a bit. |
| 21:01.28 | [TK]D-Fender | Someone's failure does not constitute a completely validated test |
| 21:01.29 | wdoekes2 | trunk chan_sip.c: check_peer_ok() => (1) sip_find_peer(of, NULL, TRUE, FINDUSERS, FALSE, 0); (2) sip_find_peer(NULL, &p->recv, TRUE, FINDPEERS, FALSE, p->socket.type); |
| 21:01.46 | wdoekes2 | (1) lookup by name, (2) lookup by IP+port |
| 21:01.50 | WIMPy | Naikrovek: That's not the way it works. You want users to authenticate on every single call. And that's what usets have to do. |
| 21:01.55 | elliot98 | [TK]D-Fender: p3nguin, you there? |
| 21:02.11 | p3nguin | Yes. |
| 21:02.12 | elliot98 | [TK]D-Fender: my tests also confirm |
| 21:02.26 | elliot98 | p3nguin: regarding the peer test...you said it was not possible to authenticate while a peer |
| 21:02.29 | [TK]D-Fender | elliot98, That club isn't restricted to 1 member... |
| 21:03.07 | p3nguin | elliot98: I don't recall doing the test as type=peer. I tested user and friend, and could complete the call. |
| 21:03.40 | elliot98 | I thought you made an account a peer and tried to call |
| 21:03.44 | elliot98 | and failed |
| 21:03.44 | wdoekes2 | [TK]D-Fender: match_auth_username works for matching multiple users behind same port+IP |
| 21:04.04 | wdoekes2 | I'd prefer that over From matching as that is sometimes used for the CLI |
| 21:04.55 | [TK]D-Fender | wdoekes2, is there a sip.conf parameter to repreent that? |
| 21:04.59 | [TK]D-Fender | represent* |
| 21:05.06 | wdoekes2 | match_auth_username=yes |
| 21:05.12 | elliot98 | [TK]D-Fender: does peer or user match first, say the IP matches for one account and username matches for another? |
| 21:05.39 | [TK]D-Fender | elliot98, if all you use is peers for phone that is the issue you typically run into. |
| 21:05.40 | elliot98 | wdoekes2: that is a new feature? |
| 21:05.44 | p3nguin | user does matching based on username, peer does matching based on IP/port. |
| 21:05.45 | wdoekes2 | since 1.6 |
| 21:06.00 | elliot98 | p3nguin: but which one happens first? |
| 21:06.06 | [TK]D-Fender | wdoekes2, Good to know. I'll see about testing that later. |
| 21:06.06 | wdoekes2 | (1)! |
| 21:06.07 | p3nguin | There is no first. |
| 21:06.25 | wdoekes2 | yes there is |
| 21:06.29 | elliot98 | p3nguin: say I have two accounts, one has host="ipaddr" and one with username="username" |
| 21:06.32 | p3nguin | Oh, you mean like in the case where both can match? |
| 21:06.38 | elliot98 | p3nguin: yes |
| 21:06.39 | [TK]D-Fender | IIRC if a user match is possible that's what it'll hit first |
| 21:06.40 | p3nguin | I think username will win. |
| 21:06.54 | wdoekes2 | let me paste some code |
| 21:06.58 | wdoekes2 | 22:01 < wdoekes2> trunk chan_sip.c: check_peer_ok() => (1) sip_find_peer(of, NULL, TRUE, FINDUSERS, FALSE, 0); (2) sip_find_peer(NULL, &p->recv, TRUE, FINDPEERS, FALSE, |
| 21:07.33 | p3nguin | So username does win when both cases are true. |
| 21:08.08 | elliot98 | this is a disadvantage, because a provider, which usually does have authenticaing, should never be a friend. |
| 21:08.43 | elliot98 | sorry, which usually DOES NOT have authentication |
| 21:08.50 | wdoekes2 | everyone has auth |
| 21:08.53 | wdoekes2 | or should have |
| 21:09.05 | [TK]D-Fender | many providers don't |
| 21:09.11 | [TK]D-Fender | and that's the way it is |
| 21:09.24 | wdoekes2 | peers can get disabled auth with insecure=invite |
| 21:09.25 | p3nguin | Is that a result of SER? |
| 21:09.33 | wdoekes2 | users can get disabled auth with empty password |
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| 21:09.55 | elliot98 | providers usually do not, so they should never set be as a friend, lest they get recognized as user and won't let the call through |
| 21:10.56 | p3nguin | I've used type=user for a provider on more than one occasion. Calls come in just fine on some, and fail on others. |
| 21:11.07 | elliot98 | and as wdoekes2 says, if there is no password in the user account, no authentication is needed, which can be a real security issue |
| 21:11.22 | elliot98 | p3nguin: did you set a password for the user? |
| 21:11.37 | p3nguin | In some cases yes, others no. |
| 21:11.51 | p3nguin | Some of my ITSPs do not have a username. |
| 21:11.59 | wdoekes2 | elliot98: it all depends on what you know your provider does.. if you know it always sends from the same IP+port and no one else is on that. use type=peer |
| 21:12.22 | elliot98 | p3nguin: because if you did not, as well were doing tests before, anyone could use the username to place calls |
| 21:12.23 | wdoekes2 | if it has auth, but can connect from multiple IPs, you can use type=user |
| 21:12.39 | elliot98 | p3nguin: at least it seems according to some pleminary tests |
| 21:13.20 | wdoekes2 | elliot98: note that username=xyz does not do what you might think. it matches on the [username] |
| 21:13.59 | elliot98 | wdoekes2: it would still be possible to place calls |
| 21:14.03 | jeffspeff | wdoekes2, then what is the username=xyz for? |
| 21:14.17 | [TK]D-Fender | jeffspeff, because [this] is taken |
| 21:14.37 | [TK]D-Fender | jeffspeff, and in my dilplan I'd rather have SIP/myprovider than SIP/hjigasasd78787asdhjasj |
| 21:14.39 | wdoekes2 | jeffspeff: see sip.conf.sample, look for defaultuser= |
| 21:15.23 | elliot98 | [TK]D-Fender: is it SIP/[this] or SIP/defaultuser? |
| 21:15.37 | jeffspeff | elliot98, SIP/[this] |
| 21:15.51 | elliot98 | so defaultuser is for authentication purposes |
| 21:15.59 | elliot98 | not for matching |
| 21:16.05 | wdoekes2 | correct |
| 21:16.19 | p3nguin | ; Note: The parameter "username" is not the username and in most cases is |
| 21:16.19 | p3nguin | ; not needed at all. |
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| 21:16.53 | elliot98 | so when is it needed? |
| 21:17.06 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 21:17.25 | wdoekes2 | are you going to ask that for all options? ;) |
| 21:17.36 | p3nguin | If you need to authenticate to another peer, that's the username you'll be using to auth against the other peer. |
| 21:17.42 | *** join/#asterisk uxos (~uxos@187.164.83.236) |
| 21:18.19 | p3nguin | In the case of having to auth calls to your ITSP, your peer for it may be named [myitsp], but your username will be something like defaultuser=ef3424ffdf44ffg4 |
| 21:19.05 | [TK]D-Fender | <wdoekes2> are you going to ask that for all options? ;) <- believable. |
| 21:19.13 | p3nguin | When a call comes in from that peer, they will not have configured your account to send calls as 'myitsp' to you. |
| 21:19.49 | [TK]D-Fender | So far this topic has been stilling in "Ttheoretical Land" without a clear problem to actually solve as far back as I cared to scroll. |
| 21:20.04 | p3nguin | Don't bother scrolling further. |
| 21:20.24 | p3nguin | It has been what if this and what if that. |
| 21:20.33 | WIMPy | We could save the log as base for some good documentation. |
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| 21:20.58 | elliot98 | p3nguin: but the password works for both incoming and outgoing? |
| 21:21.51 | p3nguin | I bet they won't be sending a password, either. |
| 21:22.08 | p3nguin | So they won't send a password, and they won't send as 'myitsp' ... |
| 21:22.10 | p3nguin | What's left? |
| 21:22.14 | elliot98 | p3nguin: IP |
| 21:22.15 | p3nguin | Match on IP/port. |
| 21:22.18 | p3nguin | Exactly. |
| 21:22.29 | p3nguin | And insecure=invite. |
| 21:22.44 | elliot98 | p3nguin: but the password would be used if needed...say if I would set up a voip client as a peer |
| 21:23.00 | elliot98 | and it tries to register |
| 21:23.10 | p3nguin | If your device is registering, it'll need to use that password that you set. |
| 21:23.22 | [TK]D-Fender | elliot98, Ok, I'm not sure it's sinking in. Let try again. Registering has nothing to do with authing calls |
| 21:23.25 | elliot98 | p3nguin: and will also use that password for outgoing |
| 21:23.40 | [TK]D-Fender | elliot98, Its to tell the other side where you are. I doesn't give them a free-pass. |
| 21:23.44 | [TK]D-Fender | It* |
| 21:24.00 | p3nguin | I've never seen asterisk authenticate a call going TO a phone. |
| 21:24.04 | *** join/#asterisk patrickod_ (~patrick@79.97.57.109) |
| 21:24.12 | elliot98 | I meant not if asterisk is registering, but a remote device is trying to register with your asterisk servfer |
| 21:24.27 | p3nguin | [myphone] |
| 21:24.34 | p3nguin | secret=superduperpasswd |
| 21:24.55 | elliot98 | p3nguin: if I were to set that up as type=peer |
| 21:24.59 | p3nguin | That allows the phone to register using a user ID of 'myphone' and a password of 'superduperpasswd' |
| 21:25.18 | p3nguin | Oh yeah... I forgot type=peer in that. |
| 21:25.37 | p3nguin | [myphone] |
| 21:25.40 | p3nguin | secret=superduperpasswd |
| 21:25.43 | p3nguin | type=peer |
| 21:25.46 | p3nguin | There. |
| 21:26.05 | p3nguin | phone registers using a user ID of 'myphone' and a password of 'superduperpasswd' |
| 21:26.25 | p3nguin | Now that it is registered, asterisk knows where to send calls which are intended for that device. |
| 21:26.47 | [TK]D-Fender | checkout time, BBIAB |
| 21:27.16 | *** join/#asterisk justdave_ (~dave@unaffiliated/justdave) |
| 21:27.18 | p3nguin | I probably should also define host=dynamic, so that it is allowed to register. |
| 21:27.34 | patrickod_ | is there a way to to a config-file check in asterisk without reloading ? |
| 21:28.09 | leifmadsen | patrickod_: there is not |
| 21:28.22 | leifmadsen | that was a point of discussion at AstriDevCon this year though |
| 21:28.33 | leifmadsen | patrickod_: at this point, the only way to test is on a separate development system |
| 21:29.29 | r0m|u | that would be a good future addition for asterisk :) |
| 21:29.32 | elliot98 | p3nguin: and if that [myphone] peer asks asterisk to authenticate when aterisk sends a call to it, which usernmae/password will it use? |
| 21:30.17 | p3nguin | I've never seen asterisk authenticate a call TO a phone, so I don't know. |
| 21:30.23 | wdoekes2 | p3nguin: authing to a phone is quite easy.. I allow the customers to do auth if they want (your basic linksys does support it) |
| 21:30.43 | elliot98 | so wdoekes2, where does it take the username/password from? |
| 21:31.16 | wdoekes2 | defaultuser and secret from [myphone] |
| 21:31.42 | p3nguin | For a phone set as type=peer, I never define a defaultuser for it. |
| 21:32.06 | elliot98 | so defaultuser is when asterisk is contacting the phone and [myphone] is for when the phone is calling asterisk |
| 21:32.10 | elliot98 | and the secret is for both |
| 21:33.10 | p3nguin | In the case of an ITSP as type=peer, I define the defaultuser so asterisk can use it to auth calls to the ITSP. |
| 21:33.29 | elliot98 | p3nguin: and it takes the password from secret |
| 21:33.51 | elliot98 | so essentially, the secret would be used for both directions during authentication |
| 21:34.08 | elliot98 | in the even authentication is needed in both directions |
| 21:34.15 | p3nguin | Since a phone and an ITSP are really not that much different as far as asterisk is concerned, if you needed to auth to a phone, it would be the same as authing to the provider. |
| 21:34.17 | elliot98 | *event |
| 21:34.53 | p3nguin | Asterisk just knows of these peers as being a device which is not itself. |
| 21:35.02 | patrickod_ | leifmadsen: ah that's a shame. |
| 21:35.24 | patrickod_ | I'm trying to write a config-management script and I'd love to have a sanity check at the end of each change before they're committed |
| 21:35.47 | patrickod_ | does this code exist in one part of the asterisk source or it it split across the various modules ? |
| 21:35.56 | elliot98 | p3nguin: but the authentication is in both directions: phone/ITSP -> asterisk PBX (uses [myphone]/secret) and asterisk PBX -> phone/ITSP (uses defaultuser/secret) |
| 21:36.09 | elliot98 | so it both situations, secret is used |
| 21:36.57 | *** join/#asterisk fireman_biff (~biff@65.48.133.103) |
| 21:36.58 | WIMPy | patrickod_: Or not at all. |
| 21:37.17 | leifmadsen | patrickod_: the code you're talking about doesn't exist at all |
| 21:37.39 | patrickod_ | leifmadsen: WIMPy hmm ok. |
| 21:38.14 | wdoekes2 | elliot98: you're repeating yourself |
| 21:38.46 | elliot98 | wdoekes2: yes I am...because I just need a little clarification |
| 21:38.58 | WIMPy | elliot98: Are you going to write a summary? |
| 21:39.02 | wdoekes2 | hehe :) |
| 21:39.20 | elliot98 | WIMPy: abstract w/ footnotes and all :) |
| 21:39.35 | fireman_biff | I have a working DUNDi peer on the other side of an IPSEC tunnel. When I configure it to use the location's external IP address instead, the peer goes offline although the firewall before the PBX shows that the traffic is being forwarded to the PBX, and iptables on the PBX is set to accept everything. Any ideas? |
| 21:39.38 | p3nguin | elliot98: Unless I don't understand it myself, [myphone] is the username required by the device when talking to asterisk; the defaultuser is the value of username that is sent by asterisk when talking to the device in question. |
| 21:39.53 | elliot98 | p3nguin: yes and secret is used for both |
| 21:40.05 | elliot98 | p3nguin: in the event a password is needed |
| 21:40.10 | wdoekes2 | correct p3nguin, until the auth= parameter comes into play |
| 21:40.25 | p3nguin | which I do not use, but should look at anyway. |
| 21:41.49 | elliot98 | wdoekes2: ok, so if there is no auth=, then it uses defaultuser/secret |
| 21:41.57 | elliot98 | wdoekes2: otherwise uses auth= |
| 21:42.31 | wdoekes2 | well.. auth uses realm sent by the auth-requesting party |
| 21:42.42 | wdoekes2 | so only if the realm matches, is it used |
| 21:42.53 | elliot98 | wdoekes2: gotcha |
| 21:44.26 | fireman_biff | should dundi work through nat? |
| 21:44.36 | elliot98 | but for voip clients, the realm would be whatever is set in the phone |
| 21:44.52 | elliot98 | so defaultuser really has no use |
| 21:44.59 | elliot98 | if an account is set up correctly |
| 21:45.03 | WIMPy | fireman_biff: If you have the port forwarded. |
| 21:45.08 | elliot98 | with proper realm athentication |
| 21:45.52 | elliot98 | but that's why I guess it's called "defaultuser" when no realm matches |
| 21:45.57 | p3nguin | Where is authentication by realm usually used? |
| 21:46.14 | fireman_biff | WIMPy: apart from setting up the forward and changing the "host" in dundi.conf, does anything else need to change for me to switch from an internal ip (ipsec) to an external ip? |
| 21:46.58 | elliot98 | p3nguin: probably not very often, but it just seems that that is how the ones who came up with SIP protocol would have liked it to be |
| 21:47.05 | WIMPy | fireman_biff: You are aware that dundi and the call taht will be set up because of it need not go the same way? |
| 21:48.04 | elliot98 | so just to summarize, clients should probably be friends, so they can also accept phone calls as well, but ITSP should only peers |
| 21:48.06 | fireman_biff | WIMPy: yeah, but right now i'm not even thinking about the call, just trying to get the peers to see each other again |
| 21:48.16 | p3nguin | elliot98: Define "clients." |
| 21:48.24 | *** part/#asterisk libryder (~david@209.33.214.243) |
| 21:48.24 | fireman_biff | the dundi peers |
| 21:48.41 | elliot98 | let's define clients as the phone within the PBX |
| 21:48.45 | wdoekes2 | p3nguin: technically you could have a chain of proxies all requiring authentication |
| 21:49.20 | wdoekes2 | .. or you could have a sip proxy that chooses the right itsp by did, where some or all of them require auth |
| 21:49.24 | WIMPy | fireman_biff: Ok, so setting the hosts and forwarding the ports shoulbe be it. |
| 21:49.31 | p3nguin | Since asterisk is both a client and a server, and phones are both a client and a server, the term "client" isn't something I use very often. |
| 21:49.53 | elliot98 | p3nguin: ok |
| 21:50.02 | p3nguin | So let's just say phone when we mean phone. |
| 21:50.13 | elliot98 | p3nguin: so phones should be set up as friends and providers as peers |
| 21:50.25 | p3nguin | I set phones as type=peer in most cases. |
| 21:50.39 | fireman_biff | WIMPy: damn, thats what I thought... the firewall says its forwarding the packets but the peer still isn't coming online |
| 21:50.40 | wdoekes2 | elliot98: once again, only if you want phones to call without registering first |
| 21:50.48 | fireman_biff | WIMPy: its only UDP on 4520 right? |
| 21:50.58 | p3nguin | but friend is well-suited to a phone. |
| 21:51.15 | wdoekes2 | (which could be if they switch IP and/or port more often than they register) |
| 21:51.28 | elliot98 | but say a few phones are behind NAT, would setting things up as peers confuse things? |
| 21:51.39 | WIMPy | fireman_biff: By default, yes |
| 21:52.00 | WIMPy | fireman_biff: Do you have a bindaddr set? |
| 21:52.27 | elliot98 | wdoekes2: so if they are registering, make them a friend, since peers do not register |
| 21:52.28 | fireman_biff | WIMPy: 0.0.0.0 |
| 21:52.42 | p3nguin | If you don't also configure a different port for each phone's client, then using type=peer in that case could present a problem. |
| 21:52.44 | wdoekes2 | elliot98: wrong |
| 21:52.46 | elliot98 | howerver, peers do seem register |
| 21:52.57 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:53.22 | p3nguin | A friend is both a user and a peer. |
| 21:53.42 | wdoekes2 | 22:50 < elliot98> p3nguin: so phones should be set up as friends [i.e. also user] 22:50 < wdoekes2> [...] only if you want phones to call without registering first |
| 21:54.05 | elliot98 | it's the without registering first that's important |
| 21:54.15 | wdoekes2 | if you cannot rely on the phones to register OR if you cannot rely on the IP+port to match when they do call THEN you need type=user |
| 21:54.45 | p3nguin | Asterisk does not care if a device is registered or not, calls are still allowed as long as any required authentication is performed by the device sending the call. |
| 21:55.11 | elliot98 | so essentially, peers would be good enough. Now you say that once a phone is registered, it follows the IP/port that is registered with, so if various phones are behind NAT, there won't be a conflict unless insecure=port is set |
| 21:55.39 | p3nguin | (1552.42) <p3nguin> If you don't also configure a different port for each phone's client, then using type=peer in that case could present a problem. |
| 21:55.59 | elliot98 | p3nguin: ok, good |
| 21:56.00 | wdoekes2 | elliot98: if you require auth, and you do, you can use match_auth_username to match the peers |
| 21:56.05 | p3nguin | Example: I have two phones behind one NAT. Both phones are set as type=peer. |
| 21:56.29 | elliot98 | wdoekes2: match_auth_username instead of [myphone] |
| 21:56.36 | wdoekes2 | no |
| 21:56.38 | p3nguin | One's client uses the standard port, the other users another port, which I configure on the device and on the entry in sip.conf. |
| 21:57.12 | elliot98 | wdoekes2: so instead of what? |
| 21:57.13 | wdoekes2 | match_auth_username=yes causes asterisk to do type=user matching based on the Authorization: header username-part instead of on the From: username |
| 21:57.46 | elliot98 | wdoekes2: so callers can still use their callerid |
| 21:57.58 | elliot98 | wdoekes2: with another username |
| 21:58.04 | wdoekes2 | yes, but they should set fromuser=your_username anyway |
| 21:58.13 | wdoekes2 | and use sendrpid=yes (or pai) |
| 21:59.20 | elliot98 | sendrpid is remote-party-id, how does that translate into callerid? |
| 21:59.25 | wdoekes2 | but some can't, and in that case they can safely use the From for the cli |
| 21:59.29 | WIMPy | Since when does Asterisk support pai? |
| 21:59.52 | p3nguin | fromuser is used for asterisk to be able to send a call as a specific user name when the From: field is not the real user name, right? |
| 22:00.21 | wdoekes2 | WIMPy: hm.. trunk doesn't (according to sip.conf.sample) |
| 22:00.43 | wdoekes2 | (so you'd have to write your own dialplan matching.. prefer rpid :P ) |
| 22:00.44 | WIMPy | Ok, that was my last information as well. |
| 22:00.52 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:01.15 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:01.19 | wdoekes2 | I had seen it, but it was sendrpid only, not trustrpid |
| 22:02.08 | wdoekes2 | *gone* |
| 22:05.38 | Igneous | In a queue, is there any way to execute an agi script (or macro) right *before* a call is delivered to an agent? I know I can use membermacro or specify an agi script in cmd Queue().. but I'm trying to manipulate the CID name, which is obviously needs to be done before the channel starts ringing :( |
| 22:07.02 | p3nguin | Use a local channel as the queue member, and alter the caller id name before the dial. |
| 22:07.19 | Igneous | facedesks |
| 22:07.23 | Igneous | why didn't I think of that? |
| 22:07.30 | Igneous | thank you p3nguin.. |
| 22:07.42 | r0m|u | p3nguin, in case of nating is port forwarind required 100% of the time? |
| 22:07.44 | WIMPy | Or before calling Queue? |
| 22:07.52 | p3nguin | r0m|u: no |
| 22:08.05 | p3nguin | I have phones behind NAT and there is no port forwarding done there. |
| 22:08.06 | WIMPy | r0m|u: Where? |
| 22:08.55 | r0m|u | curiosity. people who use m0n0wall reuire not port forwarding where pfsense does require it. so I am confue... It leads me to belive pfsense is broken. |
| 22:09.04 | r0m|u | require no* |
| 22:09.27 | p3nguin | Which firewall does m0n0wall use? I thought it was also using pf. |
| 22:09.34 | r0m|u | Yes. |
| 22:09.45 | r0m|u | both are thats why I think pfsense is broken |
| 22:09.52 | p3nguin | So then they can be configured the same and work the same. |
| 22:09.58 | WIMPy | Or more secure. |
| 22:10.00 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 22:10.21 | r0m|u | both system out of a default install m0n0wall requires no port forwarding and pfsense does |
| 22:10.23 | p3nguin | I'm sure there is a difference in the set of rules one is using. |
| 22:10.35 | r0m|u | could be |
| 22:10.40 | p3nguin | List the rules and compare. |
| 22:10.54 | r0m|u | I sall. |
| 22:10.56 | r0m|u | shall* |
| 22:10.59 | r0m|u | Thanks guys |
| 22:12.10 | r0m|u | I think I found my answer. Monowall uses cone, pfsense uses symmetric for there NAT technology |
| 22:12.20 | p3nguin | Output the rules. |
| 22:12.27 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176139514.dsl.bell.ca) |
| 22:12.50 | r0m|u | p3nguin, I will. I have to re setup pfsense |
| 22:12.52 | p3nguin | The nat rules should be configured in pf.conf regardless of the type of nat that is built. |
| 22:13.31 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 22:14.17 | n3hxs | If you make changes to NAT, you should check the rules to make sure they changed too. Often they don't and have to be manually updated in pfSense. |
| 22:16.03 | r0m|u | n3hxs, I am aware of this issue and I think it was addressed in 2.0. Thanks though. I am talking about default installs. |
| 22:16.10 | r0m|u | I need to compare rules. |
| 22:16.27 | r0m|u | that* |
| 22:17.54 | *** part/#asterisk Poincare (~jefffnode@dst.ampersant.be) |
| 22:18.09 | *** join/#asterisk garymc (~chatzilla@host86-176-88-100.range86-176.btcentralplus.com) |
| 22:25.59 | n3hxs | OK, I haven't had to work on any of our upgraded units. We now have three of the 32 sites up to version 2.0 |
| 22:30.41 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 22:36.53 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 22:36.57 | elliot98 | is there any sort of "call-waiting" response in SIP? |
| 22:38.01 | elliot98 | in other words, the phone is not in busy state and can accept phone calls, but needs to tell the provider to supply an alternate ringtone |
| 22:42.42 | _Corey_ | elliot98: It would respond with a 183 session progress and provide the alternate ringtone rtp |
| 22:43.00 | _Corey_ | (not that I'm aware of an implementation like that on a phone) |
| 22:48.50 | r0m|u | n3hxs, nice! |
| 22:50.53 | *** part/#asterisk fireman_biff (~biff@65.48.133.103) |
| 23:00.21 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ilvcviyruvyxdnek) |
| 23:06.09 | *** join/#asterisk jeffspeff (~jeffspeff@173-11-144-149-houston.txt.hfc.comcastbusiness.net) |
| 23:06.41 | jeffspeff | is anybody here familiar with MALLOC_DEBUG? |
| 23:09.08 | citywok | I appear to be stuck... any suggestions? csgtacsip1*CLI> module reload chan_sip.so -- Reloading module 'chan_sip.so' (Session Initiation Protocol (SIP)) Previous SIP reload not yet done |
| 23:09.25 | citywok | all sip traffic is pretty much dead, phones can't register, can't make new calls but existing calls are still alive |
| 23:09.42 | jeffspeff | citywok, try just "reload" |
| 23:09.43 | p3nguin | sip reload does what? |
| 23:09.47 | jeffspeff | and reload everything |
| 23:10.07 | WIMPy | bets you need a kill -9. |
| 23:10.09 | citywok | jeffspeff: in 1.6.2.20 i don't have just "reload" |
| 23:10.17 | citywok | WIMPy: yea me too but i'm really hoping not to drop an important call |
| 23:10.18 | citywok | lol |
| 23:10.21 | jeffspeff | citywok, core reload |
| 23:10.22 | WIMPy | core reload ... |
| 23:10.30 | p3nguin | And sip reload does what? |
| 23:10.34 | dijib | core reload |
| 23:10.35 | jeffspeff | lol |
| 23:10.37 | dijib | ooops |
| 23:10.42 | dijib | hey all |
| 23:10.45 | citywok | csgtacsip1*CLI> core reload No such command 'core reload' (type 'core show help core reload' for other possible commands) |
| 23:10.51 | citywok | sip reload just reloads sip.conf |
| 23:11.03 | p3nguin | That's not good enough right now? |
| 23:11.13 | citywok | it can't because it appears to be stuck |
| 23:11.19 | p3nguin | okay |
| 23:11.21 | WIMPy | Seen the "..."? |
| 23:11.22 | citywok | when i do sip reload it says it can't do it b/c it's alrady trying |
| 23:11.39 | jeffspeff | citywok, sounds like you need to restart asterisk then |
| 23:11.42 | citywok | "Previous SIP reload not yet done" -- i'm guessing that's why all my phones are now no service |
| 23:11.45 | p3nguin | So it must be trying. |
| 23:12.02 | p3nguin | You'll have to wait for the other calls to die, I'd guess. |
| 23:12.24 | p3nguin | This is when I would consider core restart gracefully. |
| 23:12.27 | citywok | yea that's my thought, but i don't see a way out of it, and i have a call going that i can't kill |
| 23:12.31 | dijib | core restart gracefully |
| 23:12.44 | p3nguin | ... and just wait. |
| 23:12.45 | dijib | :) |
| 23:12.47 | citywok | my guess is it wouldn't restart |
| 23:12.50 | citywok | (on a graceful) |
| 23:12.59 | dijib | y? |
| 23:13.00 | citywok | if it is stuck reloading sip it will probably think it isn't idle / can't restart |
| 23:13.21 | WIMPy | Most definitely. |
| 23:13.26 | citywok | core restart would likely work, but i doubt graceful will (but i'll test it in 15 minutes and let you know) |
| 23:13.52 | citywok | it's either core restart or kill-9. |
| 23:13.53 | p3nguin | gracefully doesn't check channel statistics? |
| 23:14.13 | citywok | it does check channel stats to see if tehre are any open calls before restarting |
| 23:14.23 | citywok | but i'm guessing it checks to make sure there aren't any other locks |
| 23:14.24 | p3nguin | So when the channels go away, it'll restart. |
| 23:14.40 | WIMPy | Once you tried graceful or when convenitent, there won't be a way around -9. |
| 23:14.44 | dijib | core stop gracefully - This command prevents new calls from starting up in Asterisk, but allows calls in progress to continue. When all the calls have finished, Asterisk stops. |
| 23:15.03 | citywok | yea, new calls can't originate right now anyways haha |
| 23:17.36 | r0m|u | waz up dijib |
| 23:17.43 | dijib | not much you? |
| 23:17.51 | dijib | wondering what else this asterisk box can do for me |
| 23:17.55 | r0m|u | chillin at worl fixing to got to class |
| 23:18.26 | dijib | what r u learning in class? |
| 23:18.41 | r0m|u | going for an MBA right now. |
| 23:18.50 | dijib | oh geez i didnt know it was you |
| 23:18.53 | dijib | you change your name too much |
| 23:18.57 | r0m|u | rofl |
| 23:19.02 | dijib | :) |
| 23:19.07 | dijib | wish i had me a job |
| 23:19.09 | r0m|u | all ways have been r0m|u at work :) |
| 23:19.30 | dijib | whats your name normally? Seri? |
| 23:19.35 | r0m|u | yes |
| 23:19.40 | dijib | k |
| 23:20.10 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 23:20.32 | dijib | im bored |
| 23:20.47 | WIMPy | Fix some bugs |
| 23:20.50 | r0m|u | I have to get my English at par though. Right now it sucks... lol so taking some business grammar |
| 23:21.50 | r0m|u | dijib, I am at your chan |
| 23:22.16 | citywok | It wouldn't even core restart (i never tried gracefully) |
| 23:22.19 | dijib | joining |
| 23:22.20 | citywok | i had to kill-9 it |
| 23:22.27 | hardwire | mooooooo |
| 23:24.13 | dijib | dude asterisk just crashed on me |
| 23:24.15 | dijib | wft |
| 23:24.22 | r0m|u | that sucks |
| 23:24.34 | r0m|u | logs! |
| 23:24.42 | dijib | uhm... yeh.... |
| 23:24.56 | *** part/#asterisk hacim (~micah@debian/developer/micah) |
| 23:25.19 | citywok | i had a network interface die in one of my Hyper-V cluster nodes (1 of 5), which owned a volume and caused all VM's on that volume to crash, which cascaded and crashed my entire cluster. |
| 23:25.26 | citywok | an hour later my phone system decided to go TU |
| 23:25.52 | citywok | and had to leave my entire call center offline for 20 minutes for a call to finish that was more important than the call center. |
| 23:26.03 | citywok | stupid monday |
| 23:26.07 | r0m|u | citywok, you didnt have it redundant |
| 23:26.33 | r0m|u | well what I mean is that your cluster does notmove your vms around in a predictive failure? |
| 23:26.52 | r0m|u | well your nic died so wouldnt matter |
| 23:26.53 | citywok | the Hyper-V network interfaces are not redundant MPIO, we had to replace some backend switching last month before we could do that. |
| 23:26.54 | r0m|u | that sucks |
| 23:27.13 | r0m|u | sorry! bad monday indeed |
| 23:27.13 | citywok | that's going to be happening in the near future, which would hopefully prevent this from happening again |
| 23:27.31 | r0m|u | citywok, I am using RHEL6 with LVM clustering |
| 23:27.42 | r0m|u | for our dom0 |
| 23:27.51 | citywok | we're a msft partner and "get" to use msft products |
| 23:27.58 | citywok | on the bright side it's free :-\ |
| 23:28.16 | r0m|u | it allows me to move images in case of failures since the vm sits on the lvm cluster |
| 23:28.38 | r0m|u | to prevent exactly what happen to you |
| 23:28.42 | citywok | yea i can move them around all i want unless the volume itself crashes b/c of a nic failure :( |
| 23:29.08 | citywok | even though every server has direct FC access to the volume apparently they still rely on the volume's availability on the network as well |
| 23:29.09 | r0m|u | shivers. |
| 23:29.14 | r0m|u | Thats a nasty thought |
| 23:29.44 | dijib | heh |
| 23:29.46 | citywok | yea... bad day |
| 23:29.50 | r0m|u | dijib, you back up? |
| 23:29.55 | dijib | blame bill gates |
| 23:29.59 | r0m|u | lol |
| 23:29.59 | dijib | yeh it is |
| 23:30.42 | r0m|u | I am in |
| 23:31.24 | dijib | something it crashing it |
| 23:31.48 | dijib | i just tried to join again from originating the call from the command line |
| 23:31.49 | r0m|u | :/ |
| 23:31.52 | dijib | but it crashed. |
| 23:31.57 | dijib | now im worried |
| 23:31.59 | jeffspeff | is anybody here familiar with MALLOC_DEBUG? |
| 23:32.56 | citywok | jeffspeff: i'd check in #asterisk-dev |
| 23:34.09 | *** join/#asterisk Ionic (ionic@ionic.de) |
| 23:34.15 | r0m|u | dijib, enable core debug and try again and see what spills out |
| 23:34.23 | carrar | probably OWNED!! |
| 23:34.32 | r0m|u | p2wn3d! |
| 23:34.38 | r0m|u | LOL |
| 23:34.55 | r0m|u | carrar, polycom haz u! |
| 23:35.35 | carrar | I do have a bright orange shirt on |
| 23:35.48 | r0m|u | the phone is working beautifully and the boot is 10 times faster when using split :) |
| 23:36.17 | carrar | TRUELY AMAZING!! |
| 23:36.18 | r0m|u | I have everything setup via ftp. Thanks for the help. |
| 23:36.31 | carrar | np |
| 23:36.53 | r0m|u | It all makes since when you read the admins manual :) |
| 23:37.06 | carrar | That is crazy how that works |
| 23:37.22 | r0m|u | I am still puzzled |
| 23:37.37 | r0m|u | how is that :? |
| 23:37.39 | r0m|u | lol |
| 23:38.14 | carrar | something about cell storage in the brain I think |
| 23:38.23 | r0m|u | wow |
| 23:39.51 | r0m|u | dijib, what you found? |
| 23:41.46 | r0m|u | did you found a txt file call anonymous.txt? in /etc/asterisk/? |
| 23:42.27 | r0m|u | crank up your inner wincsp powers! |
| 23:42.30 | r0m|u | :P |
| 23:42.35 | r0m|u | bored |
| 23:44.05 | hardwire | BOOOOOORED |
| 23:44.08 | hardwire | do work |
| 23:44.20 | *** join/#asterisk blerp (~blerp@S0106000c42bcfc93.cg.shawcable.net) |
| 23:44.21 | r0m|u | done with work :) now waiting for class :P |
| 23:45.21 | r0m|u | is reading The Book |
| 23:45.36 | *** join/#asterisk luckyaba (~Lucky@ip72-194-218-169.sb.sd.cox.net) |
| 23:47.23 | *** join/#asterisk Wiretap_Work (~wiretap@unaffiliated/wiretap) |
| 23:47.49 | Igneous | frowns |
| 23:48.23 | Igneous | anyone have any idea why ${QEHOLDTIME} doesn't seem to be set unless it's called by membermacro? |
| 23:49.03 | Igneous | Even if I use get_full_variable from the AGI and reference the exact channel, it still comes back empty. |
| 23:50.42 | *** join/#asterisk F2Knight (~ben@c-24-22-60-186.hsd1.or.comcast.net) |
| 23:50.47 | F2Knight | Hello everyone |
| 23:53.10 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |